[Asterisk-Users] IAX2 Destroying channel to avoid deadlock
I am receiving this message intermittently. It is happening during call setup. My phone is registering correctly. I am also having this problem between Asterisks.Any ideas where this comes from? Jun 29 01:05:04 NOTICE[13192]: chan_iax2.c:1601 iax2_destroy: Avoiding IAX destroy deadlock Jordan Novak Communications Technician ___ --Bandwidth 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 patch
On Wed, 2006-06-28 at 23:00 -0500, Aaron Daniel wrote: If anyone's interested, I've just put together a sip realtime patch, figured anyone that uses realtime may want to have a look at it. The patch basically takes the stuff asterisk updates (fullcontact, ipaddr, port, regseconds, and username) out of the sippeers table and puts it in it's own table. For those that are using multiple tables, this allows you to create a view of those tables that munges it together in a manner that makes sense to asterisk, while alleviating some of the management from you, as well as letting you make a table structure that makes sense to you ;) http://bugs.digium.com/view.php?id=7443 The one on the bugs site is for SVN, but I do have a version that works on 1.2.? (only tried on 1.2.9.1, so it may work on older versions). So far it seems to work well. If I get some spare time I wouldn't mind playing around with the patch for 1.2.9.1. Can you please stick that one on bugs.digium.com too. 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] 2 or more ISDN cards: which comes first ??
Hello, I have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM Softphone on windows 2000
Hello, I'm currently testing the SNOM softphone for one of our clients. Is anyone on this list using this software on Windows 2000 as a normal user? When we configure the softphone as an administrator and restart the software, the configured values stay the same. But when we configure it as a normal user, all values are resettet after restarting the software. This only happens when we use Win2k not with XP. Thanks for any help or hint. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth 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] ASTCC: customer wants 100 accounts
Ronald Wiplinger wrote: He want to use 100 phones at the same time!!! Alas, he won't be able to. Re: ASTCC in-use flag You'll have to disable the in-use flag for his account. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth 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 and Debian on Intel Macmini
2006/6/28, Tzafrir Cohen [EMAIL PROTECTED]: The absense of USB?Use kernel 2.6?--Tzafrir Cohensip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406[EMAIL PROTECTED]http://www.xorcom.comHi, There's no PCI slot expansion on Intel Macmini.It's possible to install Debian and Asterisk on such a platform.My question is would it be then possible to benefit from every Asterisk feature which are known to be ztdummy dependant like IVR, conference calls, ... My understanding is that you need to have a zaptel hardware to run ztdummy so, as you can't get any PCI card inside a Macmini, it's not possible to have these features.Is it correct to think so ? Regards ___ --Bandwidth 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] isdn-data over iax
Hi, [EMAIL PROTECTED] wrote: is the following zaptel.conf configuration correct for TDMoE used for pri-cpe signalling - is this possible at all ? I couldn't find an example... Any kind of Zaptel signalling should be fine. Check out http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE Best regards, Florian ___ --Bandwidth 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] Trixbox maunual configuration
One quote from the Melb Asterisk users group - Trixbox is great if you like learning things twice. (once the gui way, then once the right way.) PaulH -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8100 mob: 0434 673 529 shadowym wrote: Just my opinion and I am no expert. I took the time to crawl first. I installed basic Linux then I installed basic Asterisk. After spending some time crawling with this setup I set up some extensions and made some calls and watched the messages on the CLI while Asterisk did it's thing. After spending some time walking I installed FreePBX and then Trixbox. Now I feel fairly comfortable running FreePBX and making manual changes via the x_custom.conf abilities in FreePBX. It is trying to be all things to all people so bloat is inevitable. So far I have not found any limits that cannot be overcome by manual changes to x_custom.conf files in FreePBX. I suspect if someone just jumps in with both feet and runs Trixbox without learning the basics first they are asking for trouble when they want to get serious about it and run into the inevitable snags. Lot's of people are probably doing it this way and working backwards which is quite possible but is probably a much more difficult way to go about it. My 2 cents. *From:* Mimmus [mailto:[EMAIL PROTECTED] *Sent:* Wednesday, June 28, 2006 6:05 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* RE: [Asterisk-Users] Trixbox maunual configuration I can confirm this. AMP/TrixBox is a wonderful project but if you like to tweak something or you became a more experienced user, it will became soon as a straitjacket. I'm still struggling to clean AMP config files to work with a plain Asterisk install. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jordan Novak *Sent:* Wednesday, June 28, 2006 2:24 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Trixbox maunual configuration I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to do everything with. Trying to edit the configs manually proves impossible due to the excessive use of includes and macros. It is kind of like watching someone try to bite their own ear off. Has anybody tried to wipe all the configs clean and program the switch manually. Will this interfere with the other apps. I would wipe out extensions.conf, voicemail.conf, IAX.conf SIP.conf queues.conf and agents.conf. I do not want to use the FreePBX again after this. I am not trying to put down FreePBX, I know a lot of people have worked very hard on this. It just over complicates things for me. Jordan Novak Communications Technician ___ --Bandwidth 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 A200 hangup detection
Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? I got * server 1.2.5 and running on Centos 4.3, I hock up to two PSTN lines and each time some one call in and my phone delay 1-2 sec (this is Asterisk delay nothing to do with Sangoma) and it rings on my phone, however, end on the day I got not less that 10 empty messages. I found out that Sangoma FXO port's does not hangup the lines after the external caller hangup the trunk line's. It take about 30sec later.so bad. I did feedback to Sangoma about this and never one of the tries successescan some one help me on this? Or this is the nature of Sangoma A2XX card? I did tried with TDM4XX no hangup issues on FXO port. My zaptel.conf file fxsks=3-4 ...I did tried out ls b4 I ask Sangoma loadzone=sg Zapata.conf [channels] context = from-pstn3 switchtype = national usecallerid=yes hidecallerid=no transfer=yes echocancel = yes echocancelwhenbridged = yes echotrainning = yes busydetect=yes busycount=1 callprogress=yes relaxdtmf=yes rxgain =-2.5 txgain =-2.5 signalling=fxs_ks group=1 channel=3-4 Any advice? ___ --Bandwidth 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] 2 or more ISDN cards: which comes first ??
On Thu, June 29, 2006 8:40, Stefan-Michael. Guenther (in-put GbR) said: Hello, I have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, lspci should tell you... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth 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] using kannel with asterisk
hello I have an asterisk server with a te110pE1 digium card. the server is a hp ML370 3,2 Ghz 64bits, 1Mo L2 , 1Go Ram, 3 SCSI 73Go in raid5. I want to use in the same machine the kannel SMSC. i have no big trafic in the two gateway but I want to know if it generate a performence problem for asterisk I use fedora core4 with latest asterisk version . thanks Regards issam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 hangup detection
chan (Alpha Trilogies Networks) wrote: Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? snip [channels] context = from-pstn3 switchtype = national usecallerid=yes hidecallerid=no transfer=yes echocancel = yes echocancelwhenbridged = yes echotrainning = yes busydetect=yes busycount=1 callprogress=yes relaxdtmf=yes rxgain =-2.5 txgain =-2.5 signalling=fxs_ks group=1 channel=3-4 Any advice? A couple of things: 1. The switchtype setting is only for PRI lines. 2. Try setting callprogress=no, call progress analysis is supposedly only valid in the US. 3. Tune your gain settings until you get an optimal signal level -- google the list or the Wiki, it's quite thoroughly documented. Flynn ___ --Bandwidth 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] Call length limitation
Hi I am sending the results of my research to the list. Unfortunately any combination of hangupcause worked :(.But I also try this L option on other machine, on one of my Zap channels and this time L worked perfectly. The channel went to hangup state and Asterisk executed the DeadAGI. So I guess the L option failure is the issue of my VoIP service provider. It doesn't solve the problem but there is little I can do about itThanks for helpCheersAndrew ___ --Bandwidth 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] 2 or more ISDN cards: which comes first ??
On Thu, 29 Jun 2006, Francesco Peeters wrote: On Thu, June 29, 2006 8:40, Stefan-Michael. Guenther (in-put GbR) said: Hello, I have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, lspci should tell you... It depends on the order you load these cards. The first card loaded will be capi controller 1. 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] bristuff hangup issue
hi, Just wanted to inform everyone, if you're using the latest bristuff's you might (depends on the country!) have hangup issues. The issue appears every time you dial an external number, and hangup after letting it ring for a few times. Then the remote party keeps ringing. In some situations (we only encountered this while dialing to other * servers) it keeps the line open on the telco-side. Meaning.. you pay for it! The cdr on the calling asterisk (with the bug) doesn't indicate a long connection time. However, the cdr on the called asterisk does.. (I've seen several durations of over 20 hours) A show channels doesn't indicate any active calls. A quick fix has been posted a while ago by Marcel van der Boom (in libpri/q931.c), this works. According to the release notes this should have been applied to the latest bristuff, but be careful, the problem still exists on bristuff-0.3.0-PRE-1q. I have emailed junghanns.net to let them know. Best regards, stoffell ___ --Bandwidth 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] WIFI sip phone
UT Starcom T1000. Ive tested it in a LAN environment and its cheap and easy to configure, gives a great sound quality and the roaming behavior is pretty correct In a WAN env we havent tested it . Regards Josep De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Alessio Focardi Enviado el: miércoles, 28 de junio de 2006 18:57 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] WIFI sip phone Hi folks! Based upon your experience on the field what wifi sip phone would you reccomend ? A customer asked for a wireless * install and I'm looking for advice, tnx Alessio Focardi [[*] - Interconnessioni Italy ___ Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. This message is intended exclusively for its addressee and may contain information that is CONFIDENTIAL and protected by professional privilege. If you are not the intended recipient you are hereby notified that any dissemination, copy or disclosure of this communication is strictly prohibited by law. If this message has been received in error, please immediately notify us via e-mail and delete 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
Re: [Asterisk-Users] bristuff hangup issue
2006/6/29, stoffell [EMAIL PROTECTED]: I have emailed junghanns.net to let them know.Did they acknowledge the issue ?The issue appears every time you dial an external number, and hangup after letting it ring for a few times.Is it really every time ?Cheers ___ --Bandwidth 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 with Sipbroker calling / routing problem
Hello all, I've been using * for quite some time and yesterday I decided to add sipbroker to my config. It was pretty simple and it works for some numbers (e.g. I can call *258-9123, UK date time - which is on the phone numbers you can call page -) but fails for some others. For example I've got a friend who's at freephonie so to call him, I would dial *759608 (7596 being freephonie.net). When I do that, I get the following error: Jun 29 10:27:21 NOTICE[7916]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as32d2cdfe' And here's a snippet of what I get from 'sip debug': -- -- SIP read from 24.196.79.163:5060: SIP/2.0 407 authentication required Allow: UPDATE,REFER Call-ID: [EMAIL PROTECTED] Contact: sip:212.27.52.5:5060 CSeq: 102 INVITE From: sip:[EMAIL PROTECTED];tag=as32d2cdfe Proxy-Authenticate: Digest realm=freephonie.net,nonce=012dd3995b84e8f56ca34a7201a0c6ff,opaque=012daad2220ed2c,stale=false,algorithm=MD5 Record-Route: sip:24.196.79.163;lr;ftag=as32d2cdfe Server: Cirpack/v4.40 (gw_sip) To: sip:[EMAIL PROTECTED];tag=01-08146-012dd3ab-3b2383163 Via: SIP/2.0/UDP 172.16.1.1:5060;received=86.216.233.69;rport=5060;branch=z9hG4bK76bd560d Content-Length: 0 --- (12 headers 0 lines)--- Transmitting (no NAT) to 24.196.79.163:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK76bd560d;rport From: sip:[EMAIL PROTECTED];tag=as32d2cdfe To: sip:[EMAIL PROTECTED];tag=01-08146-012dd3ab-3b2383163 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 29 10:27:21 NOTICE[7916]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as32d2cdfe' Transmitting (NAT) to 172.16.1.19:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.16.1.19:5060;branch=z9hG4bK9954d222975cdcc1;received=172.16.1.19 From: sip:[EMAIL PROTECTED];user=phone;tag=2858979361 To: sip:[EMAIL PROTECTED];user=phone;tag=as4eecd6f3 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing -- Here's what I got in sip.conf for sipbroker: [sipbroker-out] type=peer fromuser=0001 fromdomain=somehost.somedomain.tdl host=sipbroker.com port=5060 canreinvite=yes qualify=yes Any idea what's going on? I've been reading quite a few papers about SIP authentication but I still fail to understand what's really happening (or is freephonie not 'open')? Any help is welcome! Cheers, -- Mathieu Chouquet-Stringer [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] recommended telephones
Hello all i wondered what telephones should you recommend to use with asterisk, sip compatible, that could use as many functions as possible, like any modern digital phone with programable keys. It should have leds that display who is busy at the moment, let transfer calls as simple as possible, display who is calling, allow multiconference... Where can i get that information?? Thanks and pardon my bad English Ricardo. ___ --Bandwidth 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 A200 Caller ID in UK
Hi, can anybody confirm if there are any patches required for Caller ID to work on a Sangoma A200 card on a BT line in the UK? With Asterisk 1.2.9.1 thanks 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] Suggested Phone
On 28 Jun 2006, at 23:36, Corporate IT Solutions - Michael Dunne wrote: If price is an issue, then Grandstream is the go. If quality is the issue, then Snom or Cisco. I like the elmeg 290 - nice feel to the phone and not too expensive. 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
Re: [Asterisk-Users] bristuff hangup issue
On 6/29/06, Olivier [EMAIL PROTECTED] wrote: I have emailed junghanns.net to let them know. Did they acknowledge the issue ? I didn't get any reply yet. (but I'm used to that ;)) But yes, the -q release CHANGES file contained this: - libpri fix for P2P BRI in Belgium But the bug still exists (at least in Belgium on ISDN), before it also existed in The Netherlands if I'm correct. The issue appears every time you dial an external number, and hangup after letting it ring for a few times. Is it really every time ? Yes, on calling cell phones, other * servers, fixed PSTN's, .. Just want to inform and see if anyone else is 'infected' :) , we had an extra bill of 300€ last month due to this error. (in the beginning we didn't even knew the error was there..) Cheers ___ --Bandwidth 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 SIP Registrations
On 29 Jun 2006, at 02:08, Aaron Daniel wrote: Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate tables with config information, and use a view for the actual compiled configuration, instead of how it is now, where there may be repeating info all over the database. Does any of that make sense? Yes, except, if I understand you correctly, you would also need to write insert and update triggers on the view, so that when asterisk writes to the compiled config, the correct changes are applied to your separate tables. That might limit your choice of databases a bit. The other thing to watch is that you have to ensure that the resulting view behaves exactly the way that asterisk expects it to, unless you get the join right, you can get duplicate (apparently identical) records back which would confuse asterisk. Overall I like the idea, we do this sort of thing lots in the web world, I'll probably try something similar in cdr odbc . By the way, has anyone used cdr_odbc to oracle XE (the free one) yet ? Tim. -- Aaron 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 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
Re: [Asterisk-Users] Standard Sound Files Distortion
On 28 Jun 2006, at 19:50, Douglas Garstang wrote: -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 28, 2006 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Standard Sound Files Distortion Douglas Garstang wrote: I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. What I've learned from reading the list, is it usually is a sign of shared IRQs. Just a thought. Thanks for the reply. I just worked out what it was. I had ulaw copies of all the sound files in the digits/ directory. For some reason, the ulaw files either had the cracks and pops in the recordings, or when asterisk played the ulaw files, it generated the cracks and pops. I've noticed something that may (or may not) be related. If you have a sound file that isn't an exact multiple of 20ms long, then asterisk 1.2.9.1 (don't know about other versions - yet) sends out a 'partial' packet with the remaining data in it. For ulaw, the data would normally be 160 bytes, but a few (the last?) packet(s) might be 54 bytes (or whatever). This confuses my softphone. Do we think this is correct behavior ? Shouldn't asterisk pad the sound out to 20ms? Note - this never occurs with GSM data, like the 'standard' voice files, as gsm is always a multiple of 20ms long. T. 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
Re: [Asterisk-Users] Suggested Phone
On 6/28/06, Forrest Beck [EMAIL PROTECTED] wrote: So far we have a Grandstream 2000 Cisco 7912 Very good phone but not so big display. Polycom SoundPoint IP What model? they recently released an alternative to the 501, being a 430. Looks promising. And we are looking at getting a Linksys SPA-942 My current price-wise favorite is the thomson st2030, good hardware quality for a decent price. Better then GXP-2000. (combine a plantronics headset with the st2030 or a polycom, that's all you'll ever need ;-)) cheers ___ --Bandwidth 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] Very bad quality with AVM Fritz!card PCI and chan_capi
Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear scratches and plops. It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to play with echotail and echosquelch but the quality is always terrible. Any suggestion is welcomed. Thanks, Ben /etc/asterisk/capi.conf [general] rxgain=0.5 txgain=0.5 language=fr ;set default language ;ulaw=yes;set this, if you live in u-law world instead of a-law [ISDN1] isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial) ;when using NT-mode, 'DID' should be set in any case incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any controller=1 ;capi controller number to use group=9 ;dialout group softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection accountcode= ;Asterisk accountcode to use in CDRs context=capi-in ;context for incoming calls ;echosquelch=2 ;_VERY_PRIMITIVE_ echo suppression echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;(possible values: 'no', 'yes', 'force', 'g164', 'g165') echocancelold=no ;use facility selector 6 instead of correct 8 (necessary for older eicon drivers) echotail=128 ;echo cancel tail setting devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) ___ --Bandwidth 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 card A101 Card troubles.
Hiya all, I have had no end of trouble trying to get my A101 E1 card working on a new asterisk installation. The sangoma tech people have ignored my emails about this. All the installation of wanpipe seems to go ok, and zaptel. it all installed compiled and does all the wanpipe hwprobe exactly as documented. Asterisk compiled ok, but when I run it give me Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10614 setup_zap: Unknown signalling method 'pri_cpe' Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10239 setup_zap: Signalling must be specified before any channels are. Am I right in thinking that's it's something to do with libpri? Mark ___ --Bandwidth 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] app_sms not working anymore
Hi, I have been using app_sms for a few weeks now, since I recently upgrade to asterisk 1.2.9.1 (latest bristuff, -q) however, app_sms doesn't seem to work that well anymore.. On receiving an sms, I execute the app_sms script, and get this as output: -- Accepting voice call from '171701' to 'ournumber' on channel 0/1, span 2 -- Executing Goto(Zap/4-1, custom-smsrx|sms|1) in new stack -- Goto (custom-smsrx,sms,1) -- Executing SMS(Zap/4-1, asterisk-20020-1151573913.346|a) in new stack -- SMS TX 93 00 6D -- Hungup 'Zap/4-1' For some reason there's no SMS RX after the TX, can this be a bug, or is this telco related? cheers ___ --Bandwidth 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 and Debian on Intel Macmini
On Thu, Jun 29, 2006 at 08:56:05AM +0200, Olivier wrote: 2006/6/28, Tzafrir Cohen [EMAIL PROTECTED]: The absense of USB? Use kernel 2.6? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com Hi, There's no PCI slot expansion on Intel Macmini. It's possible to install Debian and Asterisk on such a platform. As for Debian: I figure Etch will do. Don't know about Sarge, if this is too non-standard a platform. As for lack of PCI: there is some non-PCI Asterisk hardware. Heck, I work for a company that makes one. And although the Digium S100U is discontinued, I heard someone on the dev list writing a zaptel driver to use some USB modem for certain USB modems. My question is would it be then possible to benefit from every Asterisk feature which are known to be ztdummy dependant like IVR, conference calls, ... ztdummy should allow you just that: zaptel timing based on the system's clock, which can be used in the absense of zaptel hardware. -- 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] Very bad quality with AVM Fritz!card PCI and chan_capi
On Thu, 29 Jun 2006, Benjamin Sebbah wrote: Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear scratches and plops. It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to play with echotail and echosquelch but the quality is always terrible. Any suggestion is welcomed. echotail is for hardare-echo-cancel only (e.g. Eicon Diva Server) and does not change anything for AVM card. Did you try to disable echosquelch? Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I use no gain by setting it to 1.0, which works here good. Armin Thanks, Ben /etc/asterisk/capi.conf [general] rxgain=0.5 txgain=0.5 language=fr ;set default language ;ulaw=yes;set this, if you live in u-law world instead of a-law [ISDN1] isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial) ;when using NT-mode, 'DID' should be set in any case incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any controller=1 ;capi controller number to use group=9 ;dialout group softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection accountcode= ;Asterisk accountcode to use in CDRs context=capi-in ;context for incoming calls ;echosquelch=2 ;_VERY_PRIMITIVE_ echo suppression echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;(possible values: 'no', 'yes', 'force', 'g164', 'g165') echocancelold=no ;use facility selector 6 instead of correct 8 (necessary for older eicon drivers) echotail=128 ;echo cancel tail setting devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) ___ --Bandwidth 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] No Sounds
Hi @ all, after installing and compiling Asterisk there is a strange error. No sounds are played. There is a log entry, e.g. Playing 'vm-intro' (language 'en') but nothing happened. asterisk-sounds-1.0.9 is allready installed. Can you help me? Thanks and greets, Boerni ___ --Bandwidth 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] Very bad quality with AVM Fritz!card PCI andchan_capi
Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I use no gain by setting it to 1.0, which works here good. Does anyone know if you need to set rx/txgain to 0.0 to disable gain... or it is a percent value... DV ___ --Bandwidth 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] using kannel with asterisk
Well kannel by itself doesen't use much resources as far as I remember.. it's all about actions taken upon receiving sms.. Please let me know your experiences since I'm also interested in kannel / asterisk combination.. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: Thursday, June 29, 2006 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] using kannel with asterisk hello I have an asterisk server with a te110pE1 digium card. the server is a hp ML370 3,2 Ghz 64bits, 1Mo L2 , 1Go Ram, 3 SCSI 73Go in raid5. I want to use in the same machine the kannel SMSC. i have no big trafic in the two gateway but I want to know if it generate a performence problem for asterisk I use fedora core4 with latest asterisk version . thanks Regards issam __ NOD32 1.1632 (20060629) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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] Sangoma card A101 Card troubles.
On 6/29/06, Mark Ackroyd [EMAIL PROTECTED] wrote: Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10614 setup_zap: Unknown signalling method 'pri_cpe' Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10239 setup_zap: Signalling must be specified before any channels are. Am I right in thinking that's it's something to do with libpri? You probably did not build/install libpri before building asterisk, so it will be built without PRI support. Cheers, Steve ___ --Bandwidth 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] app_sms not working anymore
I thought that this was me going mad. I'm trying to use SVN trunk and have exactly the same problems. So, I think it's a bug. Julian. stoffell wrote: Hi, I have been using app_sms for a few weeks now, since I recently upgrade to asterisk 1.2.9.1 (latest bristuff, -q) however, app_sms doesn't seem to work that well anymore.. On receiving an sms, I execute the app_sms script, and get this as output: -- Accepting voice call from '171701' to 'ournumber' on channel 0/1, span 2 -- Executing Goto(Zap/4-1, custom-smsrx|sms|1) in new stack -- Goto (custom-smsrx,sms,1) -- Executing SMS(Zap/4-1, asterisk-20020-1151573913.346|a) in new stack -- SMS TX 93 00 6D -- Hungup 'Zap/4-1' For some reason there's no SMS RX after the TX, can this be a bug, or is this telco related? cheers ___ --Bandwidth 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] Very bad quality with AVM Fritz!card PCI and chan_capi
- Original Message - From: Armin Schindler [EMAIL PROTECTED] Date: Thursday, June 29, 2006 11:48 am Subject: Re: [Asterisk-Users] Very bad quality with AVM Fritz!card PCI and chan_capi On Thu, 29 Jun 2006, Benjamin Sebbah wrote: Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear scratches and plops. It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to play with echotail and echosquelch but the quality is always terrible. Any suggestion is welcomed. echotail is for hardare-echo-cancel only (e.g. Eicon Diva Server) and does not change anything for AVM card. Did you try to disable echosquelch? Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I use no gain by setting it to 1.0, which works here good. Armin Thanks for the answer, I did try to disable equosquelch but it doesn't change anything (by the way it is disabled right now). To see if the noise was a gain issue I tried to modify rx/txgain to see if it changed anything but it didn't so I let those two values at 0.5 Any other idea? Ben Thanks, Ben /etc/asterisk/capi.conf [general] rxgain=0.5 txgain=0.5 language=fr ;set default language ;ulaw=yes;set this, if you live in u-law world instead of a-law [ISDN1] isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial) ;when using NT-mode, 'DID' should be set in any case incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any controller=1 ;capi controller number to use group=9 ;dialout group softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection accountcode= ;Asterisk accountcode to use in CDRs context=capi-in ;context for incoming calls ;echosquelch=2 ;_VERY_PRIMITIVE_ echo suppression echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;(possible values: 'no', 'yes', 'force', 'g164', 'g165') echocancelold=no ;use facility selector 6 instead of correct 8 (necessary for older eicon drivers) echotail=128 ;echo cancel tail setting devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) ___ --Bandwidth 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] app_sms not working anymore
On 6/29/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I thought that this was me going mad. I'm trying to use SVN trunk and have exactly the same problems. So, I think it's a bug. Can you confirm sending out works fine? I send out an SMS without any problem, on receiving however, I have that error, and I also think the telco side thinks the delivery is okay. cheers ___ --Bandwidth 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] Issue with using dialing PBX digits after call is connected
Hi, I'm trying to make an apparently simple thing work, but I don't see how it is possible with Asterisk. This is my extensions.conf: exten = 1234,1,Dial(SIP/123456/555-555-|20|D()) ;After call connects, send DTMF exten = 1234,2,VoiceMail([EMAIL PROTECTED]); What I obviously want is that if nobody answer the call, go to voicemail. Basic stuff. Problem is Asterisk recognizes the call as being bridged as soon as the PBX on the other end answer, regarldess of whether the final extension answers or not. In other words, priority 2 will never kick in. How do I get around this? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Two FXO: How to dial a number when a RINGcomes in?
Hi,
I have tried and here it works fine (asterisk 1.2.1), with the following
configuration:
zapata.conf
context=testing
channel = 5
extensions.conf
[testing]
exten = s,1,Dial(ZAP/1/07XX)
from CLI:
-- Starting simple switch on 'Zap/5-1'
-- Executing Dial(Zap/5-1, ZAP/1/07XX) in new stack
-- Called 1/07XX
-- Zap/1-1 answered Zap/5-1
-- Attempting native bridge of Zap/5-1 and Zap/1-1
And the calls are bridged.
So, as others sugest, double check ZAP/2 channel.
BR,
Ioan - www.modulo.ro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vincent
Delporte
Sent: Thursday, June 29, 2006 1:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Two FXO: How to dial a number when a RINGcomes
in?
Hi
From: Noah Miller [EMAIL PROTECTED]
Sorry for the long delay in responding. I didn't see you message until
now due to the postfix problems on the mailing list.
No problem. I've decided to dump the rPath PoundKey linux distro because it
was still using Asterisk 1.2.5 and it was pointless to try to solve this
issue using older versions. Incidently, at least two people tried to do
what I'd like to do... but failed.
I'm beginning to think no one at Asterisk ever had the idea that maybe
someone would want to use Asterisk just as a simple bridget between two
POTS lines, with no IVR...
I think Eric Wieling is right. You have another problem not related to
what you are trying to do in the dialplan. It sounds like one of your fxo
cards or one of your phone lines is not working properly (or maybe
both). Test both phone lines and both interfaces by dialing into both of
them (make sure they are pointed to a context in the extensions.conf, and
make sure they have something to do there when you try to dial). Can you
get in to the asterisk box at all? Then
try swapping the phone lines with the fxo interfaces. Can you dial in
then?
I'll finish installing Asterisk tomorrow (got an error when compiling
Zaptel on Fedora 5, but found the probable reason why on the web forum).
Once it's up and running, I'll go through the tests, including setting up
an SIP softphone on a Windows host and trying to call out or be called in
through both FXO cards.
In the mean time, the config files are really basic:
- FILES -
ZAPTEL.CONF
fxsks=1,2
loadzone=fr
defaultzone=fr
ZAPATA.CONF
[channels]
context=cherbourg
signalling=fxs_ks
usecallerid=yes
echocancel=yes
channel=1,2
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
TRUNK=Zap/2 ; Trunk interface
[cherbourg]
;If RING on Zap/1, just dial remote through through Zap/2
exten = s,1,NoOp(Before Dialing out through ${TRUNK})
exten = s,n,Dial(${TRUNK}/01XX)
exten = s,n,NoOp(After Dialing out through ${TRUNK})
- FILES -
Your help is much appreciated :-)
VD.
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.394 / Virus Database: 268.9.5/377 - Release Date: 27/06/2006
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[Asterisk-Users] Slightly OT: SQL query to find max load
Hi, my Asterisk records CDR logs in a MySQL table. Is there anyone having a SQL query to find max load (max concurrent calls) of my system? Thanks in advance -- 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] Very bad quality with AVM Fritz!card PCI andchan_capi
On Thu, 29 Jun 2006, Mimmus wrote: Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I use no gain by setting it to 1.0, which works here good. Does anyone know if you need to set rx/txgain to 0.0 to disable gain... or it is a percent value... It's percent. Meaning: gain=1.0 leaves the voice data as is. 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] Very bad quality with AVM Fritz!card PCI and chan_capi
On Thu, 29 Jun 2006, Benjamin Sebbah wrote: - Original Message - From: Armin Schindler [EMAIL PROTECTED] Date: Thursday, June 29, 2006 11:48 am Subject: Re: [Asterisk-Users] Very bad quality with AVM Fritz!card PCI and chan_capi On Thu, 29 Jun 2006, Benjamin Sebbah wrote: Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear scratches and plops. It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to play with echotail and echosquelch but the quality is always terrible. Any suggestion is welcomed. echotail is for hardare-echo-cancel only (e.g. Eicon Diva Server) and does not change anything for AVM card. Did you try to disable echosquelch? Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I use no gain by setting it to 1.0, which works here good. Armin Thanks for the answer, I did try to disable equosquelch but it doesn't change anything (by the way it is disabled right now). To see if the noise was a gain issue I tried to modify rx/txgain to see if it changed anything but it didn't so I let those two values at 0.5 Any other idea? Maybe the card/driver has a problem. IRQ issue? Armin Thanks, Ben /etc/asterisk/capi.conf [general] rxgain=0.5 txgain=0.5 language=fr ;set default language ;ulaw=yes;set this, if you live in u-law world instead of a-law [ISDN1] isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial) ;when using NT-mode, 'DID' should be set in any case incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any controller=1 ;capi controller number to use group=9 ;dialout group softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection accountcode= ;Asterisk accountcode to use in CDRs context=capi-in ;context for incoming calls ;echosquelch=2 ;_VERY_PRIMITIVE_ echo suppression echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;(possible values: 'no', 'yes', 'force', 'g164', 'g165') echocancelold=no ;use facility selector 6 instead of correct 8 (necessary for older eicon drivers) echotail=128 ;echo cancel tail setting devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) ___ --Bandwidth 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] Re: siemens pbx and asterisk
Do I still need an ATA adapter for my analog phones once I was able to connect my Siemens HiPath 3750 to Asterisk?Thanks in advance.On 6/27/06, richard Coco [EMAIL PROTECTED] wrote: hi all,The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.I'am not sure but i thing that the feature CallerIDName was introduced in version 3 of the H.323standard. More informations about the owerviews at http://www.packetizer.com/voip/h323/.-Concerning HiPathv3.0.In version 3.0 the HiPath has a new board (the HG3540)which supports SIP (for Endpoints) and SIPQ for SIP-trunking. You are now able to interconnectAsterisk and HiPath using H.323, ISDN and/or SIPQ.rich--- Herchi Silviu [EMAIL PROTECTED] wrote: Hi, As I wrote, the HiPath needs to be upgraded to version 3 (don't ask me any details, I'm not a Siemens expert) in order to have the CallerID name passed over the H.323 link. Earlier versions (my case) ony sends and accepts the CallerId number. I have set up a workaround for calls coming to Asterisk: an AGI script sets the CallerID name according to their CallerID number by looking it up in a database. This is done in real time for every incoming call. Obviously it doesn't work for calls going from Asterisk to the HiPath. Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Michael Hamann Sent: 27 June 2006 14:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: siemens pbx and asterisk Hi Silviu, did you manage to get the callername to work? I have a comparable setup with a hipath System but I can�t get the callername to be displayed over the trunk. The callernumber works but not the name... Any suggestion? Thanks Michael We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers. On the first one we use a H.323 trunk (it needs a card on the PBX, I think it's called HG3550). It works pretty well, except for one missing feature - the callerid name is not transmitted over the link (it is a limitation of the PBX that should disappear when it is upgraded to the V3 version). The nice thing is it doesn't take any special hardware on the Asterisk server - you just have to compile and setup an H.323 channel (asterisk-oh323 works best for us). On the second one we have a Digium TE110P connected to the PBX using a PRI. It works well too, you just need the PBX to have a trunk defined and you're ready to go. We only use ten channels, so I can't say if the performance is better. In this case you need libpri and zaptel on the Asterisk. I hope this helps, Silviu --- Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito ___ --Bandwidth 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__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 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] Very bad quality with AVM Fritz!card PCI and chan_capi
On Thu, 29 Jun 2006, Benjamin Sebbah wrote: - Original Message - From: Armin Schindler [EMAIL PROTECTED] Date: Thursday, June 29, 2006 11:48 am Subject: Re: [Asterisk-Users] Very bad quality with AVM Fritz!card PCI and chan_capi On Thu, 29 Jun 2006, Benjamin Sebbah wrote: Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear scratches and plops. It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to play with echotail and echosquelch but the quality is always terrible. Any suggestion is welcomed. echotail is for hardare-echo-cancel only (e.g. Eicon Diva Server) and does not change anything for AVM card. Did you try to disable echosquelch? Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I use no gain by setting it to 1.0, which works here good. Armin Thanks for the answer, I did try to disable equosquelch but it doesn't change anything (by the way it is disabled right now). To see if the noise was a gain issue I tried to modify rx/txgain to see if it changed anything but it didn't so I let those two values at 0.5 Any other idea? Maybe the card/driver has a problem. IRQ issue? Armin That is possible, how could I check that? I can see that IRQ 17 is shared between eth0 and my fritz!card but I don't know if it changes anything: ~# cat /proc/interrupts CPU0 0: 1458224031IO-APIC-edge timer 1:953IO-APIC-edge i8042 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 14: 689691IO-APIC-edge ide0 15: 24IO-APIC-edge ide1 16: 1457845530 IO-APIC-level uhci_hcd:usb1, wctdm 17: 721402147 IO-APIC-level fcpci, eth0 18: 1457844720 IO-APIC-level uhci_hcd:usb3, wctdm 19: 0 IO-APIC-level uhci_hcd:usb2 23: 0 IO-APIC-level ehci_hcd:usb4 NMI: 0 LOC: 1458419318 ERR: 0 MIS: 0 Benjamin Thanks, Ben /etc/asterisk/capi.conf [general] rxgain=0.5 txgain=0.5 language=fr ;set default language ;ulaw=yes;set this, if you live in u-law world instead of a-law [ISDN1] isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial) ;when using NT-mode, 'DID' should be set in any case incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any controller=1 ;capi controller number to use group=9 ;dialout group softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection accountcode= ;Asterisk accountcode to use in CDRs context=capi-in ;context for incoming calls ;echosquelch=2 ;_VERY_PRIMITIVE_ echo suppression echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;(possible values: 'no', 'yes', 'force', 'g164', 'g165') echocancelold=no ;use facility selector 6 instead of correct 8 (necessary for older eicon drivers) echotail=128 ;echo cancel tail setting devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) ___ --Bandwidth 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
Re: [Asterisk-Users] app_sms not working anymore
Yeah, sending works fine. Julian. stoffell wrote: On 6/29/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I thought that this was me going mad. I'm trying to use SVN trunk and have exactly the same problems. So, I think it's a bug. Can you confirm sending out works fine? I send out an SMS without any problem, on receiving however, I have that error, and I also think the telco side thinks the delivery is okay. cheers ___ --Bandwidth 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] hipath 3750
Hello all,My Siemens PBX is hipath 3750, since HG3550 i think is applicable only to hipath 4000 for interfacing with asterisk,what do you think will I needing for asterisk and hipath 3750?Thanks. Lito ___ --Bandwidth 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 ACD with Polycom IP501
Could anybody post a working sip.cfg and phone1.cfg for the polycom IP501 thats working with the agent login. Thanks, Dean. -Original Message- From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED] Sent: 28 June 2006 17:25 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk ACD with Polycom IP501 BJ, One other thing, did I need to have a version of asterisk already installed before your version? I had a blank system with Debian installed lipri-1.2.3 (make clean, make, make install) installed zaptel-1.2.6 (as above) done svn checkout http:..functions asterisk-polycom cd into asterisk-polycom did make clean, make, make install, make samples Edited the samples to get it to work. Does that sound right? Thanks again for you help, Dean. -Original Message- From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED] Sent: 28 June 2006 11:22 To: BJ Weschke Subject: RE: [Asterisk-Users] Asterisk ACD with Polycom IP501 The show version has the following: Asterisk SVN-bweshke-polycom_acd_functions-r36151 built by I have done the sip trace, not sure if it makes a file to pull off, but the screen shows: ---(14 headers 0 lines)--- Creating new subscription sending to 192.1.3.103 :5060 (no NAT)- this is the correct IP for the phone Found peer '501' Looking for in demo (domain 192.1.3.101)- correct asterisk ip Transmitting (no NAT) to 192.1.3.103:5060: SIP/2.0 404 Not Found Hope that helps, if you need any more lines or if there is a file I can pull. Thanks, Dean. -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: 27 June 2006 12:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk ACD with Polycom IP501 On 6/27/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I'm new to this and don't know how to do a sip trace, but have attached the files as requested. Thanks for your help. Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 26 June 2006 15:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk ACD with Polycom IP501 Hi Dean - It should be working. If not, please email me a sip debug trace along with your /etc/asterisk/agents.conf and your /etc/asterisk/sip.conf. Thanks. BJ On 6/26/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: Hi, Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo (demo as from extensions.conf context) I have setup an agent in agents.conf as ,1234,Name I have changed in the sip.cfg of the polycom phone: feature.15.name=acd-login-logout feature.15.enabled=1 feature.16.name=acd-agent-availability feature.16.enabled=1 and in the phone1.cfg of the polycom I'm only using line1 so made the changes below: reg.1.acd-login-logout=1 reg.1.acd-agent-available=1 I get the login button on the phone, and when I try and login with the agent it just goes back to login. Hi. We really need a sip debug to try and capture what's happening here. Enable/Uncomment the full line in your logger.conf file and then issue sip debug from your CLI and then try your agent login again. With that, we'll be able to see behind the scenes what's going on. Additionally, please tell me what you get when you do a show version from the CLI. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Very bad quality with AVM Fritz!card PCI andchan_capi
That is possible, how could I check that? I can see that IRQ 17 is shared between eth0 and my fritz!card but I don't know if it changes anything: Can you try it (or eth0) in a different slot (or change IRQ's in the BIOS if possible) to see if it makes any difference? That's the only way to know for sure, anything else is just speculation :) James ~# cat /proc/interrupts CPU0 0: 1458224031IO-APIC-edge timer 1:953IO-APIC-edge i8042 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 14: 689691IO-APIC-edge ide0 15: 24IO-APIC-edge ide1 16: 1457845530 IO-APIC-level uhci_hcd:usb1, wctdm 17: 721402147 IO-APIC-level fcpci, eth0 18: 1457844720 IO-APIC-level uhci_hcd:usb3, wctdm 19: 0 IO-APIC-level uhci_hcd:usb2 23: 0 IO-APIC-level ehci_hcd:usb4 NMI: 0 LOC: 1458419318 ERR: 0 MIS: 0 Benjamin Thanks, Ben /etc/asterisk/capi.conf [general] rxgain=0.5 txgain=0.5 language=fr ;set default language ;ulaw=yes;set this, if you live in u-law world instead of a-law [ISDN1] isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial) ;when using NT-mode, 'DID' should be set in any case incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any controller=1 ;capi controller number to use group=9 ;dialout group softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection accountcode= ;Asterisk accountcode to use in CDRs context=capi-in ;context for incoming calls ;echosquelch=2 ;_VERY_PRIMITIVE_ echo suppression echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;(possible values: 'no', 'yes', 'force', 'g164', 'g165') echocancelold=no ;use facility selector 6 instead of correct 8 (necessary for older eicon drivers) echotail=128 ;echo cancel tail setting devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) ___ --Bandwidth 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 ___ --Bandwidth 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] hipath 3750 + hg1500 + asterisk
Has anyone here successfully tried this?hipath 3750 -- hg1500 -- asteriski'm not sure with the flowlines though.Thanks.Lito ___ --Bandwidth 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] Voicemail volume adjustment
It's not in the right syntax. Debugging the console should display that. It probably comes from my original message having the 'u' in the front, sorry about that - was in a hurry typing. For #1: - usg(2)[EMAIL PROTECTED] should be: [EMAIL PROTECTED]|usg(2) For #2: - [EMAIL PROTECTED]|g(2) should be: [EMAIL PROTECTED]|usg(2) That's weird that is causes asterisk to crash for #2 - what version of Asterisk are you running? Worse case you should just get a message saying that entry 'us1006' doesn't exist. Cullin J. Wible wrote: Because: usg(2)[EMAIL PROTECTED] causes the app to exit with a non-zero status, Because: [EMAIL PROTECTED]|g(2) causes asterisk to hard crash. And trying to use g2 in either case doesn't work either. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dustin Wildes Sent: Wednesday, June 28, 2006 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail volume adjustment Why use an application like sox - when you can make the voicemail application do it natively: exten = s,1,Dial(SIP/100,10) exten = s,2,Voicemail([EMAIL PROTECTED]|ug(10)) The key is the g(10) parameter: From the 'show application voicemail': g(#) - Use the specified amount of gain when recording the voicemail message. The units are whole-number decibels (dB). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --Bandwidth 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: *** Spam *** [Asterisk-Users] recommended telephones
Check out http://www.digium.com/en/ecosystem/partners/interoppartners.php CS From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RicardoSent: Thursday, June 29, 2006 11:16 AMTo: asterisk-usersSubject: *** Spam *** [Asterisk-Users] recommended telephones Hello alli wondered what telephones should you recommend to use with asterisk, sip compatible, that could use as many functions as possible, like any modern digital phone with programable keys. It should have leds that display who is busy at the moment, let transfer calls as simple as possible, display who is calling, allow multiconference...Where can i get that information??Thanks and pardon my bad EnglishRicardo. ___ --Bandwidth 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] Voicemail volume adjustment
Okay, that would make sense if you wanted 2 different volume levels for the messages. Just typically if the email attachment has low volume, usually the message on the phone is low too. In any case - you have 2 options now for adjusting volume. :-) Aaron Daniel wrote: The other problem is that if you add the gain to the original message, it seems to me the volume on the phone will be too loud as compared to the volume of the emailed message. Just a thought. ___ --Bandwidth 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] Very bad quality with AVM Fritz!card PCI andchan_capi
That is possible, how could I check that? I can see that IRQ 17 is shared between eth0 and my fritz!card but I don't know if it changes anything: Can you try it (or eth0) in a different slot (or change IRQ's in the BIOS if possible) to see if it makes any difference? That's the only way to know for sure, anything else is just speculation :) James Unfortunately I can't, because the LAN board is integrated and I have no PCI device left anymore. ~# cat /proc/interrupts CPU0 0: 1458224031IO-APIC-edge timer 1:953IO-APIC-edge i8042 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 14: 689691IO-APIC-edge ide0 15: 24IO-APIC-edge ide1 16: 1457845530 IO-APIC-level uhci_hcd:usb1, wctdm 17: 721402147 IO-APIC-level fcpci, eth0 18: 1457844720 IO-APIC-level uhci_hcd:usb3, wctdm 19: 0 IO-APIC-level uhci_hcd:usb2 23: 0 IO-APIC-level ehci_hcd:usb4 NMI: 0 LOC: 1458419318 ERR: 0 MIS: 0 Benjamin Thanks, Ben /etc/asterisk/capi.conf [general] rxgain=0.5 txgain=0.5 language=fr ;set default language ;ulaw=yes;set this, if you live in u-law world instead of a-law [ISDN1] isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial) ;when using NT-mode, 'DID' should be set in any case incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any controller=1 ;capi controller number to use group=9 ;dialout group softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection accountcode= ;Asterisk accountcode to use in CDRs context=capi-in ;context for incoming calls ;echosquelch=2 ;_VERY_PRIMITIVE_ echo suppression echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;(possible values: 'no', 'yes', 'force', 'g164', 'g165') echocancelold=no ;use facility selector 6 instead of correct 8 (necessary for older eicon drivers) echotail=128 ;echo cancel tail setting devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) ___ --Bandwidth 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] MixMonitor Problems
Hi, I am running * 1.2.9.1 on a server recording calls via MixMonitor. I have recorded one call which according to the cdr logs was 40 minutes, but the recording seems to stop after 22. I know this problem was fixed ages ago, but has anyone else noticed this? Any idea what could be causing it? With thanks, Wildheart ___ --Bandwidth 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] Very bad quality with AVM Fritz!card PCIandchan_capi
That is possible, how could I check that? I can see that IRQ 17 is shared between eth0 and my fritz!card but I don't know if it changes anything: Can you try it (or eth0) in a different slot (or change IRQ's in the BIOS if possible) to see if it makes any difference? That's the only way to know for sure, anything else is just speculation :) James Unfortunately I can't, because the LAN board is integrated and I have no PCI device left anymore. You seem to have 2 wctdm adapters. Can you swap one of them with the fritz card? 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
RE: [Asterisk-Users] Very bad quality with AVM Fritz!cardPCIandchan_capi
If you do not use USB then I would suggest to disable this in the bios. Henk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Harper Sent: donderdag 29 juni 2006 14:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Very bad quality with AVM Fritz!cardPCIandchan_capi That is possible, how could I check that? I can see that IRQ 17 is shared between eth0 and my fritz!card but I don't know if it changes anything: Can you try it (or eth0) in a different slot (or change IRQ's in the BIOS if possible) to see if it makes any difference? That's the only way to know for sure, anything else is just speculation :) James Unfortunately I can't, because the LAN board is integrated and I have no PCI device left anymore. You seem to have 2 wctdm adapters. Can you swap one of them with the fritz card? 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 ___ --Bandwidth 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: RE: [Asterisk-Users] Very bad quality with AVM Fritz!cardPCIandchan_capi
You seem to have 2 wctdm adapters. Can you swap one of them with the fritz card? James If you do not use USB then I would suggest to disable this in the bios. Henk I'll try the usb trick first, and then if it doesn't work I'll try to swap one of the TDM400 with the fritz. But I can't do it now because people in my company won't be able to phone while I do that, which is completely impossible. Thanks. Ben ___ --Bandwidth 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] Digium TE410P configuration to connect with CIsco 3800
Hello List, Can anyone here has a working configuration of any digium e1 card that is connected to cisco 3800. Any help will be appreciated. THanks, Lito ___ --Bandwidth 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] Digium TE410P configuration to connect with CIsco 3800
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Angelito Manansala ha scritto: Hello List, Can anyone here has a working configuration of any digium e1 card that is connected to cisco 3800. The problem is the router configuration... you need these setups to try some configuration on the Linux side. Otherwise: try a cross cable (PRI cross cable is really different) and some configuration, with E1 you have only two/four configuration possibile for the D channel, 8 if you consider also CRC. All changes are in zaptel.conf in the span line (see the documentation). This is the last configuration found for a CISCO router with E1 interface, is the SAME configuration for the E1 line directly coming from telco in italy. I made some effort to obtain the router working exactly as the telco. I dont have the router configuration. zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf signalling = pri_cpe channel = 1-15,17-31 resetinterval = never immediate=no overlapdial=yes -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEo9g33r4gvOdjXD0RApSvAKCBW7e0W7uKvbsgR9oH+PcS+J5Y6ACg04PB sGt2zlBRs/vP11FeDoCBDL0= =Lz42 -END PGP 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
[Asterisk-Users] Re: Digium TE410P configuration to connect with CIsco 3800
Thanks for your reply. here is my zapata.conf configuration [trunkgroups] [channels] context=default switchtype=national signalling=pri_cpe usecallerid=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group=1 channel = 1-15 channel = 17-31 I noticed that when i reload chan_zap.so command there is a warning like this: == Parsing '/etc/asterisk/zapata.conf': Found Jun 29 21:50:58 WARNING[739]: chan_zap.c:10886 setup_zap: Ignoring switchtype Jun 29 21:50:58 WARNING[739]: chan_zap.c:10886 setup_zap: Ignoring signalling -- Reconfigured channel 1, PRI Signalling signalling -- Reconfigured channel 2, PRI Signalling signalling -- Reconfigured channel 3, PRI Signalling signalling -- Reconfigured channel 4, PRI Signalling signalling -- Reconfigured channel 5, PRI Signalling signalling -- Reconfigured channel 6, PRI Signalling signalling -- Reconfigured channel 7, PRI Signalling signalling -- Reconfigured channel 8, PRI Signalling signalling -- Reconfigured channel 9, PRI Signalling signalling -- Reconfigured channel 10, PRI Signalling signalling -- Reconfigured channel 11, PRI Signalling signalling -- Reconfigured channel 12, PRI Signalling signalling -- Reconfigured channel 13, PRI Signalling signalling -- Reconfigured channel 14, PRI Signalling signalling -- Reconfigured channel 15, PRI Signalling signalling -- Reconfigured channel 17, PRI Signalling signalling -- Reconfigured channel 18, PRI Signalling signalling -- Reconfigured channel 19, PRI Signalling signalling -- Reconfigured channel 20, PRI Signalling signalling -- Reconfigured channel 21, PRI Signalling signalling -- Reconfigured channel 22, PRI Signalling signalling -- Reconfigured channel 23, PRI Signalling signalling -- Reconfigured channel 24, PRI Signalling signalling -- Reconfigured channel 25, PRI Signalling signalling -- Reconfigured channel 26, PRI Signalling signalling -- Reconfigured channel 27, PRI Signalling signalling -- Reconfigured channel 28, PRI Signalling signalling -- Reconfigured channel 29, PRI Signalling signalling -- Reconfigured channel 30, PRI Signalling signalling -- Reconfigured channel 31, PRI Signalling signalling Then my zaptel.conf is this loadzone = us defaultzone = us span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 # set this to 16 for E1 On 6/29/06, Massimo Nuvoli [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Angelito Manansala ha scritto: Hello List, Can anyone here has a working configuration of any digium e1 card that is connected to cisco 3800. The problem is the router configuration... you need these setups to try some configuration on the Linux side. Otherwise: try a cross cable (PRI cross cable is really different) and some configuration, with E1 you have only two/four configuration possibile for the D channel, 8 if you consider also CRC. All changes are in zaptel.conf in the span line (see the documentation). This is the last configuration found for a CISCO router with E1 interface, is the SAME configuration for the E1 line directly coming from telco in italy. I made some effort to obtain the router working exactly as the telco. I dont have the router configuration. zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf signalling = pri_cpe channel = 1-15,17-31 resetinterval = never immediate=no overlapdial=yes -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEo9g33r4gvOdjXD0RApSvAKCBW7e0W7uKvbsgR9oH+PcS+J5Y6ACg04PB sGt2zlBRs/vP11FeDoCBDL0= =Lz42 -END PGP 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 -- Lito Manansala www.voicefidelity.net Mobile: +63 906 437 0459 DID: (+63) 44 7906292 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM Softphone on windows 2000
W2K had problems with Security (Surprising huh?) You may need to grant write access for the user to the Folder where SNOM is installed. I don't think SNOM is writing to the registry if so you will need to open permissions up on those keys in the hive. ___ --Bandwidth 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] Avaya 4610sw SIP setup problem
I just tried serving the files off Apache, port 80, no change... Most parameters are taken into account by the phone, except for SIP proxy and SIP registrar... Coud someone post an excerpt from their 46xxsettings.txt where I could see the format they use? Thank you in advance, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 29 June 2006 00:33To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since. On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying "Registering" for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN " sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219 " SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--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 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
[Asterisk-Users] beronet BNS40 led blinking: not working or not connected?
Hi, I've just installed a beronet BNS40 on Asterisk 1.2.9.1. Everything seems ok, asterisk gives no error (nothing inside logs) but the 4 led on the back of the card (which is NOT connected to an ISDN line) are red and flashingwhat does it mean? Is it not properly working or it means the card is not connected to any ISDN line? The card handbook says the card has red led but not their meaning. TIA Giorgio Incantalupo ___ --Bandwidth 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] beronet BNS40 led blinking: not working or not connected?
Have you startet the asterisk allready? When i boot my machine, and dont start the astersik, the LED's keep flashing all day. (even when lines are connected) and even if /etc/init/misdn_init has been startet TIP: First connect all Lines/Phones to the card, then start asterisk. not 100% sure, but i think the card or the asterisk, or the isdn stack, will not recognize any new lines added during a running asterisk session. Giorgio Incantalupo schrieb: Hi, I've just installed a beronet BNS40 on Asterisk 1.2.9.1. Everything seems ok, asterisk gives no error (nothing inside logs) but the 4 led on the back of the card (which is NOT connected to an ISDN line) are red and flashingwhat does it mean? Is it not properly working or it means the card is not connected to any ISDN line? The card handbook says the card has red led but not their meaning. TIA Giorgio Incantalupo ___ --Bandwidth 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] SNOM Softphone on windows 2000
Well we do write to the registry... Sorry about that, but how would we otherwise store the information that is needed for the phone?! CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Thursday, June 29, 2006 4:01 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 W2K had problems with Security (Surprising huh?) You may need to grant write access for the user to the Folder where SNOM is installed. I don't think SNOM is writing to the registry if so you will need to open permissions up on those keys in the hive. ___ --Bandwidth 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 SIP Registrations
On Thu, 2006-06-29 at 10:04 +0100, Tim Panton wrote: Yes, except, if I understand you correctly, you would also need to write insert and update triggers on the view, so that when asterisk writes to the compiled config, the correct changes are applied to your separate tables. That might limit your choice of databases a bit. The way I designed the second table, you wouldn't have to update any other tables with information from the sipregs table. The only information in there is information that asterisk needs to contact phones and such. So, for example, unless you need the ip address listed somewhere else in your database, you can leave it in sipregs. The other thing to watch is that you have to ensure that the resulting view behaves exactly the way that asterisk expects it to, unless you get the join right, you can get duplicate (apparently identical) records back which would confuse asterisk. That's something that you have to be careful about anyhow :) The way I'm looking at it, you can either use a view (we use 3 different tables for actual phone configuration... so a view makes sense). Or for smaller systems, use an actual sippeers table and put the info in there. -- 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
RE: [Asterisk-Users] SNOM Softphone on windows 2000
Sorry had to jump in. I had a similar problem with Mozilla. Make sure the Users can write to the config file. I just made all the Users an Administrator at the local machine from Local Users menu, and that fixes write to issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Thursday, June 29, 2006 10:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 Well we do write to the registry... Sorry about that, but how would we otherwise store the information that is needed for the phone?! CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Thursday, June 29, 2006 4:01 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 W2K had problems with Security (Surprising huh?) You may need to grant write access for the user to the Folder where SNOM is installed. I don't think SNOM is writing to the registry if so you will need to open permissions up on those keys in the hive. ___ --Bandwidth 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] Call Queue NOT using RoundRobin ?!?
I have setup several Calling Queues, each setup with RoundRobin strategy. When I call the queue, the first member/agent phone rings. Great! I call it again, the second member/agent rings?? I thought that was the RRMemory strategy, but it seems RoundRobin is also doing it. Anyone know what I can do to my queues, in order to force each call down the ordering of my members list? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 group pickup
Hello, I have set pickupgroup and callgroup for zap, sip and iax2 devices. Everything is working good with zap and sip and between these two. Iax2 pickupgroup and callgroup seems to be broken. I cannot pickup a call to IAX2 from SIP. Is there somewhere a bug ? I am running: Asterisk 1.2.9.1 Bartosz ___ --Bandwidth 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 ACD Polycom - Please help
Could anybody post a working sip.cfg and phone1.cfg for the polycom IP501 thats working with the agent login, I need to get this sorted to go live next week. If anybody can share their experience or pointers. Thanks, Dean. -Original Message- From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED] Sent: 28 June 2006 17:25 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk ACD with Polycom IP501 BJ, One other thing, did I need to have a version of asterisk already installed before your version? I had a blank system with Debian installed lipri-1.2.3 (make clean, make, make install) installed zaptel-1.2.6 (as above) done svn checkout http:..functions asterisk-polycom cd into asterisk-polycom did make clean, make, make install, make samples Edited the samples to get it to work. Does that sound right? Thanks again for you help, Dean. -Original Message- From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED] Sent: 28 June 2006 11:22 To: BJ Weschke Subject: RE: [Asterisk-Users] Asterisk ACD with Polycom IP501 The show version has the following: Asterisk SVN-bweshke-polycom_acd_functions-r36151 built by I have done the sip trace, not sure if it makes a file to pull off, but the screen shows: ---(14 headers 0 lines)--- Creating new subscription sending to 192.1.3.103 :5060 (no NAT)- this is the correct IP for the phone Found peer '501' Looking for in demo (domain 192.1.3.101)- correct asterisk ip Transmitting (no NAT) to 192.1.3.103:5060: SIP/2.0 404 Not Found Hope that helps, if you need any more lines or if there is a file I can pull. Thanks, Dean. -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: 27 June 2006 12:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk ACD with Polycom IP501 On 6/27/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I'm new to this and don't know how to do a sip trace, but have attached the files as requested. Thanks for your help. Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 26 June 2006 15:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk ACD with Polycom IP501 Hi Dean - It should be working. If not, please email me a sip debug trace along with your /etc/asterisk/agents.conf and your /etc/asterisk/sip.conf. Thanks. BJ On 6/26/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: Hi, Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo (demo as from extensions.conf context) I have setup an agent in agents.conf as ,1234,Name I have changed in the sip.cfg of the polycom phone: feature.15.name=acd-login-logout feature.15.enabled=1 feature.16.name=acd-agent-availability feature.16.enabled=1 and in the phone1.cfg of the polycom I'm only using line1 so made the changes below: reg.1.acd-login-logout=1 reg.1.acd-agent-available=1 I get the login button on the phone, and when I try and login with the agent it just goes back to login. Hi. We really need a sip debug to try and capture what's happening here. Enable/Uncomment the full line in your logger.conf file and then issue sip debug from your CLI and then try your agent login again. With that, we'll be able to see behind the scenes what's going on. Additionally, please tell me what you get when you do a show version from the CLI. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Re: Digium TE410P configuration to connect with CIsco 3800
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Angelito Manansala ha scritto: I noticed that when i reload chan_zap.so command there is a warning like this: == Parsing '/etc/asterisk/zapata.conf': Found Jun 29 21:50:58 WARNING[739]: chan_zap.c:10886 setup_zap: Ignoring switchtype Jun 29 21:50:58 WARNING[739]: chan_zap.c:10886 setup_zap: Ignoring signalling This is normal, the only way to change switchtype and signalling is to stop and restart asterisk ;-) Bye. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEo+0S3r4gvOdjXD0RAjVCAKCtPeCDQviW7yUl+t1Jwt1L8YJBGQCeOkpC uiyXHpJ4cFe+s0IpYSKdNZM= =lt7C -END PGP 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] Realtime SIP Registrations
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 28, 2006 7:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Realtime SIP Registrations Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate tables with config information, and use a view for the actual compiled configuration, instead of how it is now, where there may be repeating info all over the database. Does any of that make sense? How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? 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] Realtime patch
On Thu, 2006-06-29 at 08:39 +0200, Patrick wrote: If I get some spare time I wouldn't mind playing around with the patch for 1.2.9.1. Can you please stick that one on bugs.digium.com too. I've uploaded the 1.2.9.1 patch as well. Let me know if you find anything I did wrong (I'm not much of a coder, so I'm sure I screwed something up). -- 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
RE: [Asterisk-Users] Realtime SIP Registrations
On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) -- 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
Re: [Asterisk-Users] SNOM Softphone on windows 2000
Couldn't you also create a separate GPO that allows for Read-Only permissions?? Just in case. RandyW SANS wrote: Sorry had to jump in. I had a similar problem with Mozilla. Make sure the Users can write to the config file. I just made all the Users an Administrator at the local machine from Local Users menu, and that fixes write to issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Christian Stredicke Sent: Thursday, June 29, 2006 10:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 Well we do write to the registry... Sorry about that, but how would we otherwise store the information that is needed for the phone?! CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Alexander Lopez Sent: Thursday, June 29, 2006 4:01 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 W2K had problems with Security (Surprising huh?) You may need to grant write access for the user to the Folder where SNOM is installed. I don't think SNOM is writing to the registry if so you will need to open permissions up on those keys in the hive. ___ --Bandwidth 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] bristuff hangup issue
A quick fix has been posted a while ago by Marcel van der Boom (in libpri/q931.c), this works. The fix works, but it creates another bug afaik. Once you apply the q31.c ourcallstate/peercallstate patch as I call it, the line gets hung up normally, but for some odd reason all the scripts in the hangup extension won't run. So be aware of that. Quote from Marcel: --- I've done a bit more testing and in our install the patch seems to cause an issue with the 'hangup' (h) extensions. We use this to convert incoming faxes to pdf and send them off through mail after the sending fax machine hangs up. The hangup extension is never reached so that bit of our dialplan didnt work anymore. Since both patched and unpatched dont work with that particular setup, there's no way (i know) to test out wether this is actually caused by the patch or not, but i thought i'd just mention it. --- Cheers, Jeroen Zwarts ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax2 group pickup
I have set pickupgroup and callgroup for zap, sip and iax2 devices. Everything is working good with zap and sip and between these two. Iax2 pickupgroup and callgroup seems to be broken. I cannot pickup a call to IAX2 from SIP. Is there somewhere a bug ? I am running: Asterisk 1.2.9.1 Never worked for me too. I'm currently using app_pickup2.c: http://linux.thorsten-knabe.de/asterisk/pickup.jsp and it works like a charm with every channel. DV ___ --Bandwidth 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 Native Sound Distortion (ulaw)
I too have noticed problems with Asterisk native sounds using ulaw on Asterisk 1.2.9.1. Don't know about other versions but it seems to work quite well in Astlinux 0.40. In theory, since I am using ulaw for SIP there is no transcoding so it is a more efficient use of CPU resources and it should sound much better in general. It does sound better except for the frequent cracles, pops, and momentary dropouts which makes it much more objectionable to listen to compared to the standard GSM files. Is there a bug report on this yet? -Original Message- From: Tim Panton [mailto:[EMAIL PROTECTED] Sent: Thursday, June 29, 2006 2:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Standard Sound Files Distortion On 28 Jun 2006, at 19:50, Douglas Garstang wrote: -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 28, 2006 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Standard Sound Files Distortion Douglas Garstang wrote: I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. What I've learned from reading the list, is it usually is a sign of shared IRQs. Just a thought. Thanks for the reply. I just worked out what it was. I had ulaw copies of all the sound files in the digits/ directory. For some reason, the ulaw files either had the cracks and pops in the recordings, or when asterisk played the ulaw files, it generated the cracks and pops. I've noticed something that may (or may not) be related. If you have a sound file that isn't an exact multiple of 20ms long, then asterisk 1.2.9.1 (don't know about other versions - yet) sends out a 'partial' packet with the remaining data in it. For ulaw, the data would normally be 160 bytes, but a few (the last?) packet(s) might be 54 bytes (or whatever). This confuses my softphone. Do we think this is correct behavior ? Shouldn't asterisk pad the sound out to 20ms? Note - this never occurs with GSM data, like the 'standard' voice files, as gsm is always a multiple of 20ms long. T. 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] GXP-2000 and transferring call directly to voicemail
Hey everyone, I was wondering if anyone is able to help me with a solution. I have a small office set up with GXP-2000 phones and the one thing I cannot get to work is them being able to transfer a caller directly to another persons voicemail. If I have a dial tone (and not on a call), I can simply type *12 to go directly into extension 12s voicemail. However, when I use the TRSNFR button for a call that is active, as soon as I hit * it returns back to the call. Also, I have most of the extensions set up as BLF on the speed dial buttons, and would love it if that could work some how to use those to select which extension to dial direct to voicemail while on a call, but call the extension while not on a call. Any ideas??? THANK YOU!!! Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma card A101 Card troubles.
Yes, You are. Libpri$ make clean Libpri$ make install Zaptel$ make Zaptel$ make install Asterisk$ make Asterisk$ make install In that order. All should be well. Ben Bawkon Varion, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Ackroyd Sent: Thursday, June 29, 2006 5:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Sangoma card A101 Card troubles. Hiya all, I have had no end of trouble trying to get my A101 E1 card working on a new asterisk installation. The sangoma tech people have ignored my emails about this. All the installation of wanpipe seems to go ok, and zaptel. it all installed compiled and does all the wanpipe hwprobe exactly as documented. Asterisk compiled ok, but when I run it give me Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10614 setup_zap: Unknown signalling method 'pri_cpe' Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10239 setup_zap: Signalling must be specified before any channels are. Am I right in thinking that's it's something to do with libpri? Mark ___ --Bandwidth 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 SIP Registrations
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) 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] Asterisk Native Sound Distortion (ulaw)
shadowym wrote: I too have noticed problems with Asterisk native sounds using ulaw on Asterisk 1.2.9.1. Don't know about other versions but it seems to work quite well in Astlinux 0.40. In theory, since I am using ulaw for SIP there is no transcoding so it is a more efficient use of CPU resources and it should sound much better in general. It does sound better except for the frequent cracles, pops, and momentary dropouts which makes it much more objectionable to listen to compared to the standard GSM files. Is there a bug report on this yet? shadowym, While I haven't noticed this myself, many people have pointed this out. I can assure you that the prompts in AstLinux 0.4 are the same native prompts provided on astlinux.org. I don't know why they seem to sound so much better in AstLinux than with standard Asterisk installs, but as I said many people have noticed this. In theory, the native sounds should sound much better no matter how you play them back. Interesting... -- Kristian Kielhofner ___ --Bandwidth 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] username in Real-time changes all the time
I cannot explain that: One of my users shows up in sip show peers as 654200/Elmit_Unl I can set it back to 654200/654200 but it will change back to 654200/Elmit_Unl Why? bye Ronald Wiplinger ___ --Bandwidth 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] beronet BNS40 led blinking: not working or not connected?
Hi Kai, when I connect the ISDN line the LED is not blinking anymore. I think it is working now. Thanks. Giorgio Incantalupo Kai Ober wrote: Have you startet the asterisk allready? When i boot my machine, and dont start the astersik, the LED's keep flashing all day. (even when lines are connected) and even if /etc/init/misdn_init has been startet TIP: First connect all Lines/Phones to the card, then start asterisk. not 100% sure, but i think the card or the asterisk, or the isdn stack, will not recognize any new lines added during a running asterisk session. Giorgio Incantalupo schrieb: Hi, I've just installed a beronet BNS40 on Asterisk 1.2.9.1. Everything seems ok, asterisk gives no error (nothing inside logs) but the 4 led on the back of the card (which is NOT connected to an ISDN line) are red and flashingwhat does it mean? Is it not properly working or it means the card is not connected to any ISDN line? The card handbook says the card has red led but not their meaning. TIA Giorgio Incantalupo ___ --Bandwidth 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] Addon-ooh323 install problem
Richard, I ran into this problem today myself I am using the latest trunk to take advantage of the Jingle support (works nicely :) ) . But I need h.323 support as well. Any suggestions or patches? Thanks, Chaim ___ --Bandwidth 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] Cisco 7905G SIP firmware needed
Hi, I bought a 7905G Cisco IP Phone and want to connect to Asterisk with SIP protocol, but can't find a way to download this protocol update from Cisco, Can anyone please help me? Support the SIP protocol also the XML applications that I can use with SCCP? What is better, try to configure 7905G as SIP or try to use SCCP with Asterisk? Best regards, Andrea Frigo [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] username in Real-time changes all the time
What kinda phone is it? That shouldn't affect the actual calls to the phone, I would expect. On Thu, 2006-06-29 at 23:49 +0800, Ronald Wiplinger wrote: I cannot explain that: One of my users shows up in sip show peers as 654200/Elmit_Unl I can set it back to 654200/654200 but it will change back to 654200/Elmit_Unl Why? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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
RE: [Asterisk-Users] bristuff hangup issue
I can also add that this happens on em_w lines as well. I've had issues where callers start getting dead air when dialing out. Talking with the phone company the lines were in an off-hook state even though Asterisk hung up the call. I done exactly as below where I hang up before the other party answers the call. I've also had where after I hang up the CLI shows the call hanging up and then another call starts, starting simple switch, as if the call was re-established but Asterisk doesn't know what to do with the call and executes the s,1,hangup() on the call. This does NOT however always hang up the call on the ATT side. The T1 still shows the call as off-hook even though it's not in use. It seems random (at least I haven't figured out the pattern yet) as to when the channel gets hung up properly on ATT's side and when it's not. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stoffell Sent: Thursday, June 29, 2006 3:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] bristuff hangup issue hi, Just wanted to inform everyone, if you're using the latest bristuff's you might (depends on the country!) have hangup issues. The issue appears every time you dial an external number, and hangup after letting it ring for a few times. Then the remote party keeps ringing. In some situations (we only encountered this while dialing to other * servers) it keeps the line open on the telco-side. Meaning.. you pay for it! The cdr on the calling asterisk (with the bug) doesn't indicate a long connection time. However, the cdr on the called asterisk does.. (I've seen several durations of over 20 hours) A show channels doesn't indicate any active calls. A quick fix has been posted a while ago by Marcel van der Boom (in libpri/q931.c), this works. According to the release notes this should have been applied to the latest bristuff, but be careful, the problem still exists on bristuff-0.3.0-PRE-1q. I have emailed junghanns.net to let them know. Best regards, stoffell ___ --Bandwidth 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] 2 or more ISDN cards: which comes first ??
Hi, Am Donnerstag, 29. Juni 2006 09:46 schrieb Francesco Peeters: On Thu, June 29, 2006 8:40, Stefan-Michael. Guenther (in-put GbR) said: Hello, I have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, lspci should tell you... That's easy, here's the output: 00:09.0 Network controller: Eicon Networks Corporation Diva Server 2FX (rev 01) 00:0a.0 Network controller: Eicon Networks Corporation Diva Server 4BRI Rev 2 (rev 01) And now let's get to the interesting and more difficult part: According to the output of lspci, the capi.conf should look like as follows, right? [isdn1] ; EICON 2 FX controller=1 group=1 devices=2 [isdn2] ; EICON 4 BRI 1st port group=2 controller=2 devices=2 [isdn3] ; EICON 4 BRI 2nd port group=2 controller=3 devices=2 [isdn4] ; EICON 4 BRI 3rd port group=2 controller=4 devices=2 [isdn5] ; EICON 4 BRI 4th port group=2 controller=5 devices=2 The 2FX is not connected to an ISDN line, all ports of the 4 BRI are connected in TE mode. I guess all the other parameters of this file aren't important for my problem. The thing is, that [isdn2] is dead, but when I configure a section [isdn6], I can use this controller for outgoing and incoming calls. There isn't a mistake in the configuration of the 4 BRI card itself otherwise I couldn't use ISDN3-ISDN6. The question is, what happened to ISDN2? Thanks for help hints. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth 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 SIP Registrations
I think lots of us know about it... We're just not sure how to go about fixing it. :-( I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying works for me have never actually tested their system in failure scenarios, or they are working in a controlled environment without NAT and such... regards, David On 6/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) 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
[Asterisk-Users] (no subject)
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RE: [Asterisk-Users] Cisco 7905G SIP firmware needed
To get the SIP firmware for these phones, you need to buy a Cisco SmartNet support contract (about $75 USD in the USA, though I've heard rumors a Europe-only contract exists for about $10 USD.) You can purchase one through most Cisco resellers. That will give you access to Cisco's download site. Configuration of these phones under SIP is not quite as straight forward as it is with SCCP, but it's manageable. SCCP does not work very well with asterisk in a large deployment setup in my experience. I had approximately 25 phones on chan_sccp and the stability was nowhere near where a commercial phone system should be (we're talking 2 or 3 crashes a week) and is still lacking features to make it useful to run an entire setup on (3 way calling, unattended transfer, etc) Unfortunately, I cannot get XML services working for the life of me under the SIP image using the 7912G phones (which are essentially just a 7905 with a built in switch.) The configuration file schema has an option for a services and directory URL, but the phone seems to ignore them. XML services on the other Cisco phones like the 7940 or 7960 work fine with the SIP image (but these phones use a totally different provisioning method than the 7905/7912.) Ultimately I'd say the recommendation comes down to how many phones we're talking about and what kind of environment. Chan_sccp seems to work fine in a small system where you're not going to be adding or removing many users, but for any system you're going to have to support, I would (and do) use SIP. -Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Frigo Sent: Thursday, June 29, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 7905G SIP firmware needed Hi, I bought a 7905G Cisco IP Phone and want to connect to Asterisk with SIP protocol, but can't find a way to download this protocol update from Cisco, Can anyone please help me? Support the SIP protocol also the XML applications that I can use with SCCP? What is better, try to configure 7905G as SIP or try to use SCCP with Asterisk? Best regards, Andrea Frigo [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 ___ --Bandwidth 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] test
- Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 16, 2006 6:47 AM Subject: Re: [Asterisk-Users] Gumstix! James Harper wrote: http://www.gumstix.com For a non-telephony (Bluetooth based) project. I'm browsing the SVN website for Gumstix and lo and behold, there is Asterisk! I'm excited. Has anyone ever tried it on a GumStix before, and if so, care to share tips? I'd not heard of these before. Do you know if a BRI adapter can be obtained for them? Kristian at Astlinux is the person to talk to about these things. I think you can find out more about many aspects of embedded Asterisk at http://www.astlinux.org B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime SIP Registrations
What I did was modify sip to update the status on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial (SIP/[EMAIL PROTECTED]:5060) Of course you need to check the status in realtime data before you dial. This allows MANY Asterisk servers to share the same SIP data.I then load balance with DNS SRV.. Yes I have tested in failover it works. I too have been told that by many that this will not work. So I keep expecting to hit some problem with it, but to date I have not... Doug From: [EMAIL PROTECTED] on behalf of David Thomas Sent: Thu 6/29/2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations I think lots of us know about it... We're just not sure how to go about fixing it. :-( I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying works for me have never actually tested their system in failure scenarios, or they are working in a controlled environment without NAT and such... regards, David On 6/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) 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 winmail.dat___ --Bandwidth 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] Call Queue NOT using RoundRobin ?!?
Welcome to my personal hell ! :)I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue.I mean When a call comes in first member of the queue is ring, then second, etcSubsequent calls take the same path: first, second and so on.Someone has suggested to use ringall with penalties (pretty esotic!) but also this is not working for the purpose. I was also told that nobody wants that (you insensitive clod!) even if this call distribution seems pretty logic in some case scenarios. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically calls are rerouted to the backup operator after ringing to the first). Hope you can find out something to share, maybe we can also launch a count us initiative :)Alessio FocardiOn 6/29/06, Aaron Paxson [EMAIL PROTECTED] wrote: I have setup several Calling Queues, each setup with RoundRobin strategy. When I call the queue, the first member/agent phone rings. Great! I call it again, the second member/agent rings?? I thought that was the RRMemory strategy, but it seems RoundRobin is also doing it. Anyone know what I can do to my queues, in order to force each call down the ordering of my members list? ___--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
[Asterisk-Users] Any one with sending and receiving Sucessfull SMS PTSN Portugal?
Hi, I'm planning to develop a solution with SMS using Asterisk within Portuguese PSTN landline. Any one has made it before? I'm looking for Telco's and details using Portugal Telecom landline. Thanks in advance, -- Best regards, Marco Mouta ___ --Bandwidth 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 SIP Registrations
can you elaborate on modify sip to update the status on the sip friends in realtimethanksOn 6/29/06, Doug G [EMAIL PROTECTED] wrote:What I did was modify sip to update the status on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial ( SIP/[EMAIL PROTECTED]:5060) Of course you need to check the status in realtime data before you dial.This allows MANY Asterisk servers to share the same SIP data.I then load balance with DNS SRV..Yes I have tested in failover it works. I too have been told that by many that this will not work.So I keep expecting to hit some problem with it, but to date I have not...Doug From: [EMAIL PROTECTED] on behalf of David ThomasSent: Thu 6/29/2006 1:05 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP RegistrationsI think lots of us know about it... We're just not sure how to goabout fixing it. :-(I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying works for me have neveractually tested their system in failure scenarios, or they are workingin a controlled environment without NAT and such... regards,DavidOn 6/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Aaron Daniel [mailto: [EMAIL PROTECTED]] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) 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 listTo 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 -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth 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 and ivr systems
Hope this could help, Please note Inband DTMF won't work unless the codec is ulaw or alaw (G711). Use out of band DTMF aka rfc2833 or info. http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode best regards, Marco Mouta ps.give me some feedback if it worked On 6/29/06, Shane [EMAIL PROTECTED] wrote: Hello, Ther's probably a simple answer to this but I've searched around and haven't located anything as yet. Is there a way to have DTMF tones passed through Asterisk without it messing with them? I am using a tdm21b card and when I call an ivr system from the telephone handset (routed over sip or iax2) such as telebanking, the ivr has trouble recognizing tones. When I tested this with a remote party, I was told tones were breaking up. For example, a long press would result in a click, some silence and a small dtmf on the remote end. Triggering a speed dial didn't go well either as he heard only a few tones. I have dtmfmode=inband in sip.conf and have tried relaxdtmf=yes in zapata.conf. I realize Asterisk does need to detect dtmf for things like call parking but can it just pass the audio to the other side with no regard for whether it's dtmf digits? IE. long press results in long tone, etc. Best, Shane -- http://www.cm.nu/~shane/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth 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] Call Queue NOT using RoundRobin ?!?
The linear function helps me too. I've built an extensive multi-queue technical support system strategy. Based on the initial queue, ALL calls goes to Tier1 first. Then, if Tier1 does not get the call (on the phone/away from desk), Tier2 should get it, so on, and so forth. In Tier1, the primary helpdesk technician (like your receptionist idea) takes ALL calls (That's what they were hired for). However, others can help out, if the pri technician is on the phone. Here's my question: If roundrobin strategy remembers the last call made, and sends the next call to the next number (and this is by design), then why on earth was the RRMemory strategy created?? Thanks for your response, Alessio. ~~Aaron - Original Message - From: Alessio Focardi To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Sent: Thursday, June 29, 2006 1:31 PM Subject: Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Welcome to my personal hell ! :)I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue.I mean When a call comes in first member of the queue is ring, then second, etcSubsequent calls take the same path: first, second and so on.Someone has suggested to use "ringall" with penalties (pretty esotic!) but also this is not working for the purpose. I was also told that "nobody wants that" (you insensitive clod!) even if this call distribution seems pretty logic in some case scenarios. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically calls are rerouted to the backup operator after ringing to the first). Hope you can find out something to share, maybe we can also launch a "count us" initiative :)Alessio Focardi On 6/29/06, Aaron Paxson [EMAIL PROTECTED] wrote: I have setup several Calling Queues, each setup with RoundRobin strategy. When I call the queue, the first member/agent phone rings. Great! I call it again, the second member/agent rings?? I thought that was the RRMemory strategy, but it seems RoundRobin is also doing it. Anyone know what I can do to my queues, in order to force each call down the ordering of my members list?___--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
[Asterisk-Users] quadBRI in bri_net mode - t3 timer expired
Hi all, i successfully connected our old PBX to an asterisk server with a junghanns quadBRI, the quadBRI ports running in bri_cpe_ptmp mode connected to the interal PBX ISDN ports. Now i tried to turn it round as our PBX depends on it for some features and changed one of the quadBRI ports to bri_net signalling and connected it to one of the external PBX ISDN ports (how do you name that in telco jargon?). The card led goes green (indicates an ISDN link), but nothing happens when i try to place a call from our PBX using the new connection (no incoming call on the *-console). Instead qozap complains every few seconds Jun 29 19:39:55 asterisk kernel: qozap: activating layer 1, span 3 Jun 29 19:39:58 asterisk kernel: qozap: t3 timer expired for span 3 Jun 29 19:39:58 asterisk kernel: qozap: not re-activating layer1 span 3 Jun 29 19:39:58 asterisk kernel: qozap: clearing alarms on span 3 What is it trying to tell me? My quadBRI doesn't do any powerfeeding, might that be a problem? - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
