[Asterisk-Users] Resolving QOS problems
I'm looking for advise on troubleshooting QOS problems. After much searching and reading online (Google, Voip-Info Wiki, etc.) I don't feel any closer to finding the right tools to solve my problem. Any info you would like to share would be much appreciated, and I'm sure the thread will server others in the future. The problem : - I'm having intermittent problems with the audio cutting out on calls. At the same time the audio problems occur, I often see these in the full log : Received iseqno 122 not within window 123-123 These range from sounding like bad cell phone calls, to the audio track cutting out in one or both directions for up to 20-30 seconds. I also see dropped calls that seem to be a result of the IAX connection going away. The environment : - I've got an * server located at a data center with good connectivity, 10 hops to my IAX provider, and ~34ms ping times. They (IAX provider) use Cogent which concerns me a bit, but I'm not ready to jump to conclusions just yet. My IP phone is connected via enhanced DSL (static addresses, no PPPoE) and I'm 12 hops away from my * server. My DSL provider has direct connectivity and peering agreements with the data center my server is located in. I've set QOS priority on the LAN port (Linksys router) the phone is connected to, and I've dropped the MTU to 576 as suggested for lower speed links. (1.5Mbs/384kbps in my case). Both these changes seemed to make an improvement over previous calls. Currently I don't believe the bulk of my problems to be between the phone and the * server. testyourvoip.com tests consistently show a 4.4 score (the maximum for ulaw) and rarely shows errors. Ulaw is the codec used for both the SIP calls and IAX trunk. What I'm looking for : -- I'm trying to determine the cause and location of the problem between my * server and the IAX provider (and possibly my IP phone), and see what if anything I can do to reduce the occurrence of these drop outs. I'm looking for a couple of things : 1. A method of monitoring RTP/IAX traffic QOS at the PBX in real time. 2. Tools that might be used to determine the location of the problem. I.E. An RTP/IAX traceroute tool. What I'm hoping to find is something that either integrates directly with *, or captures live RTP/IAX traffic and provides real time statistics on calls. What I've found : - I saw Telchemy's VQMON_EP product, but it's unclear how it would work with Asterisk. Many other companies in this market seem to leverage off of Telchemy's products. http://www.telchemy.com/partners.html http://www.voiptroubleshooter.com/tools/voiptr_tools.htm All of the products above seem to be aimed at large enterprises with deep IT pockets. I wouldn't mind ponying up a reasonable sum for a tool that does the job, but I lack the time to thoroughly evaluate everything that may be out there. I haven't found much on the open source front. I've seen Windows RTP Quality Monitor which might be useful, but it's beta and hasn't been updated in over a year. It seems to me that Ethereal might be integrated with a graphical tool, and if nothing else provide postmortem statistics on a phone call. Request for comments : -- What are people using to troubleshoot these problems? What commercial software works for you? What open source projects are you using? Do you have suggestions on projects that might be glued together to provide this functionality? Thanks in advance. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Xoops Skype module
On Mon, Sep 19, 2005 at 11:41:26AM -0400, Dean Collins wrote: Anyway long story short there is a IRC module for xoops. But I was thinking how cool would it be to have a skype module. http://www.xoops.org/modules/newbb/viewtopic.php?topic_id=40632viewmode =flatorder=ASCstart=10 Basically the 'website' could initiate a permanently open conference room for various skype users to connect to whenever they are browsing the site. I thought some 'clever' people in here currently working on the Skype API might have some thoughts? AFAIK the skype API requires (by license) an interactive skype client working unobstructed. That is, not suited for a daemon process. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hfc card unplug plug not working?
Hi, we have got same problem, sometimeit may depend from your telco due to a bad data transmission synchronization. Just leave your plug always inserted into your ISDN card (why should you unplug it??). Giorgio. [EMAIL PROTECTED] wrote: Hello, I have hfc-pci card with zaphfc driver from bristuff. card is working with asterisk, but when i unplug line cable from card, and then plug back to card in log i can see only could not create Zap channel when dialing. when I remove zaphfczaptel drivers and then modprobe them card with asterisk works again and stop working when line is unplugged (then plugged back). is it normal? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need example about sjphone with asterisk
http://www.asteriskguru.com/tutorials/sjphone_softphone.html enjoy. julien bossart wrote: Hi all, I am new to this forum.Say hello to all. I need some help to make a example using sjphone with asterisk (which will fonction as SIP server). I use Fedora core 4, asterisk release version 1.2. Can you give me sip.conf and extension.conf example? Thank you so much. Julien ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstriCon 2006 Location
How about someplace central like South Africa? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with PRI
OK, I solved the problem(s) It was a completely wrong setup from the provider. Now anything is OK zttool shows OK in the alarms asterisk works great, parameters in zaptel.conf are span=1,0,0,ccs,hdb3,yellow bchan=1-15 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 bchan=17-31 # set this to 1-15,17-31 for E1 the only little problem is a delay of about 1 second or less in voice Andrea [EMAIL PROTECTED] .it Sent by: To asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject [Asterisk-Users] problems with PRI 19/09/2005 16.05 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi, I configured an asterisk box with 1 Digium Wildcard TE110P T1/E1 Card 0 I setup the jumper in e1 position. my zaptel.conf : defaultzone=it loadzone=it #gestione PRI span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 bchan=17-31 # set this to 1-15,17-31 for E1 # Asterisk starts correctly, I see th 30 channels. Anyway I cannot put any call on the PSTN PRI Channel. Actually I don't know 1) ccs or cas ? my provider (Fastweb Italy) does not answer. 2) ami or hdb3 ? same answer (no answer) 3) crc or not ? my provider says it does not matter moreover : what is supposed to show the zttool utility ? If I push Loop nothing happens. If I run ztcfg -v i see SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) with asterisk turned off, zttool shows RED/NOP Digium Wildcard TE110P T1/E1 Card 0 with asterisk turned on, zttool shows RED Digium Wildcard TE110P T1/E1 Card 0 if i try cat /dev/zap/1 with asterisk turned off, I see a lot of raw data incoming with asterisk turned on, I see cat: /dev/zap/1: Device or
Re: [Asterisk-Users] Resolving QOS problems
Chris Miller wrote: I'm looking for advise on troubleshooting QOS problems. After much Have a look at SineStatIAX: http://www.sineapps.com/sinestatiax.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup after voicemail not detected
Hi, On my FC3 box I am having *v1.0.9. The problem is that when a user calls through POTS line and leaves a message in voicemail, the channel doesn't detect the remote hangup. After 10 seconds of remote hangup it plays messages like vm-thankyou, vm-review etc as if user is still online. After that it hangs up displaying a warning message WARNING[25623]: file.c:568 ast_readaudio_callback: Failed to write frame Can it problem related to service provider...?? Had anyone solved this kind of problem earlier or if have any idea...Please help! My extensions.conf incoming context is [incoming] exten = s,1,Answer exten = s,2,Dial(Zap/1,5,tr) exten = s,3,Voicemail([EMAIL PROTECTED]) exten = s,4,Hangup * Gurminder ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stopping retransmission on messages
Chris Miller wrote: I'm seeing a number of these logged in full while my * system is idle, but I haven't found a good description of what they mean. Can someone oblige? I have a single SIP phone registered and an IAX trunk. Sounds to me like the packets (ACKS maybe) are arriving late. Sufficiently so that Asterisk is about to retransmit the packet. However, right at the last minute it got the ACK from the last one and stopped the retransmission as it found the ACK. Just a guess mind you. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PTN calls into asterisk slow release
Can anyone please give advice how to make PTN calls that terminates on * release immediately after call end? It takes up to 3 min for a call to release on our server. Thanks! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.3/106 - Release Date: 19/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipp examples
Does anyone have an example of how to use sipp and the matching extensions.conf entries ? Many thanks. Julian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there a clever way to page a group of extensions?
I want to be able to dial a 'pager' extension from an phone on my asterisk server, and have it ring all other extensions *except* the extension from which I am calling (because call waiting is enabled on most extensions by default) - effectively giving me the ability to page all other extensions from any phone. The solutions I've come up with so far (individual contexts for each extension or customised dial strings for each extensions) are pretty gruesome. Is there a neat way of achieving this functionality? Thanks Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i4l ring indication problem, again...
I 'm experiencing the same problem and another worst than ring indicator, because i'm unable to call some numbers with automatic response system. Calling these numbers with I4L gets always busy. If you call with a cell phone, for example, the same number is free. I googled and asked on several mailing lists for weeks, and the conclusion of my search is that I4L is not to take into consideration for voice. It's seem this is caused by the way I4L manage signalling. It discards everything before a CONNECT state, so you loose ring indicator and you hear nothing from automatic response system than answers without sending the CONNECT signalling. I think me and you have no way to solve this problem using I4L. Please let me know if you find a solution, i'll do the same, but i'm moving to CAPI -- Massimo Frisoni Omadon wrote: I can't find solution anywhere. I googled and find people with the same problem but there was no answers on how to fix this. I have W6692 based PCI cards that uses hisax driver (card type=36). Card is working fine under asterisk with i4l modem driver for incoming calls. If I want to dial out using some sip phone I don't get ring indication. Phone is ringing and I hear only silence until someone answers the phone. Using r in Dial command kind of fixes this problem but not completely. Example: I call someone on mobile phone and he is busy, I want get busy right away, first i will hear ringing (until calls reach mobile phone) and then i get busy. This confuses people allot. So my question is how to get ring indication on i4l. I tried different i4l cards, some of them where usb and all have the same ring indication problem. And yes I have /etc/asterisk/indications.conf, I tried to change countries but it didn't help. Please don't tell me to try CAPI or bristuff, I know they work but I want to make this work (this small usb isdn is very cool) Is there anybody that has a working configuration for i4l that doesn't have ring indication problem (I'm using 2.4.29 kernel). Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pri gateway
Hello, I haven't solved following problem yet. I worry that: CLI pri intense debug span There is no any debug information. Does it give any idea about problem? Baris Simsek wrote: hi, my asterisk version is 1.0.9 /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 it is comfortable with Turkish Telecom. i tried before and it works. /etc/asterisk/zapata.conf [channels] switchtype=euroisdn signalling=pri_cpe context=incoming group=1 channel=1-15,17-31 Leds are lighting at start. When i run /etc/init.d/zaptel they go out. And i can see the modules are installed. and i see that, layer 1 is going up after zaptel. So i am sure there is no problem with drivers. I think it is connected to asterisk. any idea? thanks... altus wrote: what about a copy of your zapata.conf and zaptel.conf,what color is the leds On Thu, 2005-09-08 at 12:42 +0300, Baris Simsek wrote: hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts also successfully. I tested my pri cable and it works. But still my span isn't up. I don't see any error. Do you have any idea? What else i should check? Thanks. My card is 4 span Wildcard TE410P http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P # lsmod wct4xxp 106688 62 zaptel226820 129 wct4xxp # asterisk -r gw*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: Valiant E1 CB and UniCall
Is there any success in connecting Valiant E1 CB with Unicall to asterisk? any help will be appreciated, Paradise Dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connect not signalled (SIP - Zap)
Hi, I've had a strange problem several times during the last days: A call is established, both parties have audio in both directions, but asterisk is still waiting for connect. Thus after timeout (120secs) the call is terminated with either busy or no answer. This is annoying for the both parties, who are already speaking, because they get interrupted. In the cdr I can find the call with 120 secs duration, 0 secs billsec. Has anyone had a similar problem so far? Or any ideas? Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Buy a digium hardware
Assuming you can purchase online, just go to voipsupply.com. http://www.voipsupply.com/index.php?manufacturers_id=13 The switch between analog and digital makes a huge difference to port density. With an analog TDM card you can get 4 FXO/FXS ports per card. With a digital T1PRI card, you can get 4 T1 spans with 23 voice channels each. If you are going to use a lot of analog ports (more than 8) then youmay benefit from moving to a channel bank and installing a PRI card to the Asterisk box. You can find more info at... http://www.voip-info.org/ Cheers, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leopoldo Rodríguez HSent: Monday, September 19, 2005 8:24 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Buy a digium hardware Where i can buy a digium hardware TDM400P in Mexicois there a hardware with more than 4 FXS/FXO ports (8, 12, 24)? that is supported by Asterisk*RegardsLeopoldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Point to Point with Fritz Card ...
I just set up a system with two ISDN pci cards and am using mISDN, plus chan_misdn (multipoint only). It seems to work fine except for a few annoyances, as I wrote in another post. I tried to ran chan_capi, afterwords, just to check on the difference but had problems. Of course, I did not load chan_misdn and chan_capi together as they are mutually exclusive, as per docs. Have you been successful in running chan_capi using the mISDN drivers? The misdn docs say you should be able, but after trying once I would like to hear experiences on this. Chan_capi has a lot of features plus fax stuff implemented that make it interesting. DB On Tue, 2005-09-20 at 13:15 +0800, Craig Guy wrote: You will need to use the mISDN drivers - the AVM CAPI drivers will not support PTP. It is possible to use mISDN with chan_capi but chan_misdn would be easier. Craig - Original Message - From: "Joao Correia" [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, September 20, 2005 4:57 AM Subject: [Asterisk-Users] Point to Point with Fritz Card ... Hello all, Does anyone has any experience with Point to Point Fritz Card and Asterisk ? I have a BRI access Point to Multipoint working fine but I can only have 3 numbers. The phone telco said that if they change to Point to Point I can have 10 numbers. Does anyone has any experience with Point to Point ? Best regards Joao Correia ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dias Badekas [EMAIL PROTECTED] Athens International Airport ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323: New versions 0.6.7 and 0.7.3
Hello all, Updated versions of asterisk-oh323 are now available both for use with Asterisk v1-0 (version 0.6.7) and Asterisk HEAD/v1-2 (version 0.7.3). Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] General Config information
I dont want to start a RTFM thread, but can someone jsut clear this up for me. In zapata.conf I have ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=incoming signaling=v23 rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0 txgain=0 group=0 callgroup=1 pickupgroup=1 immediate=yes musiconhold=default channel =1 faxdetect=incoming ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf The context=incoming part, does this mean on that line/channel that it is to use the [incoming] block I have in my extensions.conf file? Which is just [incoming] exten = s,1,Dial(SIP/200SIP/201,20,tr) exten = s,2,Voicemail,u1000 exten = s,102,Voicemail,b1000 So what ever you set context= to has to have its own entry in the extension.conf file? Also I have faxdetect=incoming what context is this using in the extension.conf file? Thanks. faxdetect=incoming ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What hardware would you recommend?
I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT PABX's, each serving a different area. 1 of those has two BRI's and a 2 port Nova Compact PABX with DECT First step would be to set up the (*) servers and have them interconnected. When all of that works we'd go on to connect them to the ISDN and connect the existing PABX's to the servers so we can - for now - maintain the existing environment but use (*) to route traffic on a least cost basis, as well as allow SIP/IAX connections from out of office locations. The machines themselves will not pose much of a problem, but what ISDN hardware would you recommend for this? (1 site with 1 TE and 1 NT mode port, 2 sites with 2 TE and 2 NT mode ports) TIA! -- 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 sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP audio port usage
So the more reliable way to do QoS is with MAC adress and not on a port basis. Am I right ? Thanks for your help, Adrien On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Its dynamically assigned on a per-call basis. Asterisk assigns the port based on contents of rtp.conf. Remote sip phones assign port numbers based on whatever the manufacturer happened to choose (no industry standard). E.g., Cisco uses 32,768 to something around 40,000, while xlite uses something in the area of 8,000. The various manufacturers are not consistent at all. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Adrien Laurent [EMAIL PROTECTED] www.modulis.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP audio port usage
Yes, because then the MACs specified would be getting the QoS, not just certain ports. This is how I set up my customers when they have QoS available. --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Adrien Laurent -Sent: Tuesday, September 20, 2005 8:53 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] SIP audio port usage - -So the more reliable way to do QoS is with MAC adress and not -on a port basis. -Am I right ? - -Thanks for your help, - -Adrien - -On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: - - I know that SIP is using port 5060 for session -initiation, but which - port does it use for audio ? is it dynamically assigned ? - - Its dynamically assigned on a per-call basis. - - Asterisk assigns the port based on contents of rtp.conf. - - Remote sip phones assign port numbers based on whatever the - manufacturer happened to choose (no industry standard). E.g., Cisco - uses 32,768 to something around 40,000, while xlite uses -something in the area of 8,000. - The various manufacturers are not consistent at all. - - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - - --- -Adrien Laurent -[EMAIL PROTECTED] -www.modulis.ca -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Locking Up
Ok... I asked a question a few months back about a 7960 that a user claims to be shocking her in her ear from time to time. A few others indicated they had similiar issues and alot of them seemed to stem from power over ethernet. Here's what we've done... We replaced the phone, ran two new cat5 cables to a different switch, put in a power brick and disabled power over ethernet. Over the last few months, the number of incidents of her getting shocked have reduced to almost never, but the phone is displaying the same symtoms as when she was getting the shock. The phone seems to lock up. We can not establish any type of pattern as to what causes it, but here's what we do know. She can be on a call and not touch any buttons. The soft keys will blank out and she loses audio as does the person on the other end. This has happened over both Zap and Sip channels. The strange thing is that if she waits about 20 seconds, the LCD panel will sort of flash and she gets the call back!! I never see anything in the CLI that makes me think Asterisk is even aware it is happening. I've done some research and I found some people have had issues with cell phone radiation locking up or rebooting a 7960. Has anyone else experienced this? We tried removing her cell phone from the room and it doesn't seem to make any difference. We do, however, have a cell phone repeater set up, but it's closer to alot of other users than her. Anyone have any suggestions on how to debug this? Is there some type of logging meter we can buy or rent that we could stick over there and monitor the environment for a week or so? As always, thanks for the help!! Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel = Kind of solution...
Have you done any testing to see if it made any difference what type of
trunk was being used?
Darren Wiebe
[EMAIL PROTECTED]
Ricardo Poppi wrote:
Hi all.
I´ve found a kind of solution (if we can call it this way...) and Im
reporting it here to help save some lives.
Editing into astcc.cgi I found where the parameters that set 60 and 30
seconds warning were and put zeros in its place. The last two
lots-of-zeros numbers at second line. So the zap trunk code of astcc.cgi
became like that:
==
if ($res-{tech} eq Zap) {
$dialstr = Zap/$res-{path}/$phone|30|HL( . ($maxtime *
60 * 1000) . :0:0);
$res = $AGI-exec(DIAL $dialstr);
$answeredtime = $AGI-get_variable(ANSWEREDTIME);
$dialstatus = $AGI-get_variable(DIALSTATUS);
$callstart = localtime();
return $dialstatus;
}
==
And - at least until now... - everything is working fine. The credit is
being take from the cards in the right amount and no warnings are being
given when 60 and 30 seconds left. When credit finishes, the agi script
just finishes the call.
If somebody has a better way to do that, please let us know.
Rgs, Ricardo Poppi.
Mensagem Original
Assunto: ASTCC speaks and cut RTP channel
Data: Fri, 09 Sep 2005 18:09:52 -0300
De: Ricardo Poppi [EMAIL PROTECTED]
Para: asterisk-users@lists.digium.com
Hi list.
I have a fine running Ser+Asterisk environment and have just installed
ASTCC. It´s working fine either, including its caller-id authentication
feature (the one we pass the card-number as CALLERID variable and
number-to-dial as EXTEN variable).
The issue, a great one, is that when the credit is about one minute to
end, the ASTCC prompt gets into the call, says that you have one minute
left... and when it was suppose to leave and let the RTP traffic of the
original call be reestablished, it never happens. The RTP packets - I
could see that at asterisk debug screen - stop running and the call is
still signaled as active, but no media at all.
This is a serious problem I´m having and, as I could see, I´m not the
only one. Mr. Chilini reported that around jun 30th this year, as you
can see bellow: (I just added a comment at this voip-info page to see if
anyone could give some clues about that)
http://www.voip-info.org/tiki-index.php?page=ASTCCGuide#comments
Do anyone here in this list had any situation alike? Do you have any
clues do help me? (and others because it will be documented, of course).
Thanks in advance,
Ricardo Poppi.
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[Asterisk-Users] Red or Yellow alarm monitoring
Before I reinvent the wheel, is anyone implementing any monitoring of PRI (or T1) Red or Yellow alarms? I would like to get notified ASAP if this occurs. Or possibly automate the fix since service zaptel reload seems to fix my random issue. I was thinking of using tail of the full log file, but my concern is that if there is too much traffic, I will miss the alarm, or if there is too little traffic, I will keep getting notified even if it is fixed. Are there any triggers in Asterisk that can run a script if this error occurs? What are others doing for this? -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP audio port usage
But, if I have Xlite running on client PC and at the same time the user is doing FTP, both service has the same QoS treatment? Is there a way to differentiate these services besides the port? Sebastian On 9/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote: Yes, because then the MACs specified would be getting the QoS, not just certain ports. This is how I set up my customers when they have QoS available. --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Adrien Laurent -Sent: Tuesday, September 20, 2005 8:53 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] SIP audio port usage - -So the more reliable way to do QoS is with MAC adress and not -on a port basis. -Am I right ? - -Thanks for your help, - -Adrien - -On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: - - I know that SIP is using port 5060 for session -initiation, but which - port does it use for audio ? is it dynamically assigned ? - - Its dynamically assigned on a per-call basis. - - Asterisk assigns the port based on contents of rtp.conf. - - Remote sip phones assign port numbers based on whatever the - manufacturer happened to choose (no industry standard). E.g., Cisco - uses 32,768 to something around 40,000, while xlite uses -something in the area of 8,000. - The various manufacturers are not consistent at all. - - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - - --- -Adrien Laurent -[EMAIL PROTECTED] -www.modulis.ca -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] Music on Hold
Hello, What could be the problem if [EMAIL PROTECTED] is not starting mpg123 even though I did not touch the MOH-config files? There is no error message in asterisk at debug/verbose level 9. It seems asterisk doesn´t even launch mpg123, but it´s hard to say - maybe it launches it for 1 second and I just can´t see it... I have googled a lot lately but could not find any hints how to solve this problem... I am running [EMAIL PROTECTED] at version release 1.0.9. mpg123r ist the mpg123 release. What could I do? Could it be a class problem? Any help I would really appreciate. Best regards, Armin Lediger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP audio port usage
Then you'll have to make sure that other services are lower QoS. Past that, find out what port XLITE uses and then QoS that port. --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Sebastian Milioto -Sent: Tuesday, September 20, 2005 9:50 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] SIP audio port usage - -But, if I have Xlite running on client PC and at the same -time the user is doing FTP, both service has the same QoS treatment? -Is there a way to differentiate these services besides the port? - -Sebastian - - - -On 9/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote: - Yes, because then the MACs specified would be getting the QoS, not - just certain ports. This is how I set up my customers when -they have - QoS available. - - --Original Message- - -From: [EMAIL PROTECTED] - -[mailto:[EMAIL PROTECTED] On -Behalf Of Adrien - -Laurent - -Sent: Tuesday, September 20, 2005 8:53 AM - -To: Asterisk Users Mailing List - Non-Commercial Discussion - -Subject: Re: [Asterisk-Users] SIP audio port usage - - - -So the more reliable way to do QoS is with MAC adress and -not on a - -port basis. - -Am I right ? - - - -Thanks for your help, - - - -Adrien - - - -On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: - - - - I know that SIP is using port 5060 for session - -initiation, but which - - port does it use for audio ? is it dynamically assigned ? - - - - Its dynamically assigned on a per-call basis. - - - - Asterisk assigns the port based on contents of rtp.conf. - - - - Remote sip phones assign port numbers based on whatever the - - manufacturer happened to choose (no industry standard). E.g., - - Cisco uses 32,768 to something around 40,000, while xlite uses - -something in the area of 8,000. - - The various manufacturers are not consistent at all. - - - - - - - - ___ - - --Bandwidth and Colocation sponsored by Easynews.com -- - - - - Asterisk-Users mailing list - - Asterisk-Users@lists.digium.com - - http://lists.digium.com/mailman/listinfo/asterisk-users - - To UNSUBSCRIBE or update options visit: - -http://lists.digium.com/mailman/listinfo/asterisk-users - - - - - - - --- - -Adrien Laurent - -[EMAIL PROTECTED] - -www.modulis.ca - -___ - ---Bandwidth and Colocation sponsored by Easynews.com -- - - - -Asterisk-Users mailing list - -Asterisk-Users@lists.digium.com - -http://lists.digium.com/mailman/listinfo/asterisk-users - -To UNSUBSCRIBE or update options visit: - - http://lists.digium.com/mailman/listinfo/asterisk-users - - - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] using a voip cable modem
is it possible to use asterisk to do provisioning for a voip cable modem or an MTA device? If so how can this be done? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP audio port usage
So the more reliable way to do QoS is with MAC adress and not on a port basis. Am I right ? Thanks for your help, Adrien On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Its dynamically assigned on a per-call basis. Asterisk assigns the port based on contents of rtp.conf. Remote sip phones assign port numbers based on whatever the manufacturer happened to choose (no industry standard). E.g., Cisco uses 32,768 to something around 40,000, while xlite uses something in the area of 8,000. The various manufacturers are not consistent at all. A very common way of handling QoS is to rely on the TOS (Type of Service) bits located in the IP header. Those bits are set in asterisk packets via a statement like: tos=lowdelay in sip.conf and iax.conf. There are similar type parameters available in most quality sip phones. However, once the bits are set properly, its then up to your router and/or switch to queue the packets properly for transmission over the network. The majority of the soho routers and switches do not have code to actually handle that queuing, and even if you have a device that does properly handle it, the prioritization of the packets is outbound traffic only. Your internet service provider would have to do something to prioritize the inbound traffic to you, and most won't do that. In addition, the majority of the backbone Internet providers don't pay any attention to any QoS settings. The QoS parameters work very nicely in corporate networks where support personnel understand the concepts and monitor their resources, but isp's and itsp's generally don't have a clue (or don't care). There are other software packages that will help prioritize packets to/from the Internet, and most of them use some form of trickery to accomplish the goal. For example, outbound http packets are delayed allowing rtp packets to be sent without delay, resulting in a form of QoS. By delaying the http packets (sent outbound), the remote web server essentially is placed in a wait state causing it not to forward any packets to your site, resulting in a form of inbound QoS. Those types of QoS will not handle streaming packets such as those associated with listening to music or watching videos. For the most part, QoS across the Internet (regardless of whose equipment you use) is not very effective today since the majority of isp's and backbone suppliers have not implemented QoS. As one example, you could have the most expensive Cisco router on your end with properly implemented QoS prioritization, but if I sent a large number of icmp or other fake packets to your IP address, I'd consume all available bandwidth leaving your rtp packets no way to reach your site reliably. For home and small offices that rely on DSL type facilities, implementing QoS can improve the quality as generally the outbound bandwidth is significantly less then inbound bandwidth. In those cases, prioritizing outbound traffic (on the low bandwidth portion) may help, but it still won't do much for inbound traffic. The exact same issues apply regardless of whether you rely on the TOS bits or MAC address method of prioritizing traffic. The TOS bits just happens to be a far more common method. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aterisk App ICES Question
I have a question about the Asterisk Application ICES. I've got Asterisk
setup to accept a phone call and call the ICES app which sends it to an
Icecast server.
exten = 1,1,SetGroup('stream')
exten = 1,2,GetGroupCount()
exten = 1,3,Ices('contrib/${GROUPCOUNT}-ices.xml')
exten = 1,4,Hangup
Everything works fine. Unless I have more than 24 phone calls being
converted at the same time. When I try to bring up the 25 phone call the
call comes up, Asterisk answers, and begins encoding the call.
Everything is ok with that call. However, as the 25 call comes up one of
the first 24 calls ( there is no pattern ) breaks. I say breaks because
Asterisk doesn't hangup on the call, but it does stop sending data to
the ICES application, which causes a timeout on the Icecast server.
If I make direct connections from the ICES application to Icecast
everything works, I can run as many simultaneous streams as I want.
I've tried increasing the open file limit with ulimit, but it doesn't help.
Does anyone have any thoughts on why this is happening?
Thanks,
Dan
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[Asterisk-Users] Cisco Ip phones
Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap thanks Sander ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Locking Up
I do recall your postings relative to this... I've done some research and I found some people have had issues with cell phone radiation locking up or rebooting a 7960. Has anyone else experienced this? We tried removing her cell phone from the room and it doesn't seem to make any difference. We do, however, have a cell phone repeater set up, but it's closer to alot of other users than her. Anyone have any suggestions on how to debug this? Is there some type of logging meter we can buy or rent that we could stick over there and monitor the environment for a week or so? US cell phones generate such low RF power there is no way for any such equipment to impact a 7960 (and you've already proven that). The same is true with cell repeaters and/or towers. (US cell equipment is rated in terms of milliwatts of power.) The issue is most likely static electricity generated by something. Probably should focus on ruling that out as a possibility. If it is a static charge, it could originate from her clothing and sliding around in her chair, shuffling her feet on the carpet, etc. Using a static mat under her chair might help and/or some of the anti-static sprays that are available. I'd be inclined to put an AC voltmeter between a metallic part of the 7960 and a true ground point. (Getting at a metallic part of the phone might be a trick since the phone is probably UL listed and doesnt' have any exposed metallic parts. And, finding a true ground point in the office might not be easy either. In the US, the round ground prong on an AC outlet should be a true ground depending on how will the electrian did his work.) I'd also check the grounding of the rack that holds the PoE switches just to ensure the source isn't coming from that area. Since you've found ...reduced to almost never... with the changes that you've made, I'd have to suspect a real grounding problem back in your switch racks, etc. FWIW, the folks that engineer central office telephony equipment (including racks, etc) are fanatics about rack grounding for some very good reasons. Some electrians understand that very well also. Most of those issues can be diagnosed with a standard high impedance AC voltmeter available from Radio Shack. In case you're not a believer (or don't understand rack grounding), consider the case where a complete equipment rack is not grounded in any form, and one piece of equipment in the rack develops a problem where one side of the 110 volt AC commercial power is shorted to the chassis. The equipment can and will operate properly, but touching the rack and a water pipe tends to be an issue. ;) Since the equipment in such a rack is generally connected (one way or another) to other racks in the same room or building, one rack problem can certainly impact another distant rack. Been there, done that, felt it, and had to fix it for real. You shouldn't need any specialized test equipment to diagnose either a static electricity or grounding problem. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one way voice
Hi I have set up an Asterisk System with One XLite Phone and when i call the trunk line or receive calls via a trunk line (FXO generic X100P) i'm getting one way Voice. I can hear the called party - but they cannot hear me... Any ideas - is t a NAT issue or is it something to do with the generic X100P card. How does one sort this problem-- Mark D'CruzD'Cruz Consulting www.dcruz.netM: 07932 554993 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one way voice
is your asterisk server outside of your internal network? If not then nat should not be an issue and it would point to the X100P clone. On 9/20/05, Mark D'Cruz [EMAIL PROTECTED] wrote: Hi I have set up an Asterisk System with One XLite Phone and when i call the trunk line or receive calls via a trunk line (FXO generic X100P) i'm getting one way Voice. I can hear the called party - but they cannot hear me... Any ideas - is t a NAT issue or is it something to do with the generic X100P card. How does one sort this problem -- Mark D'Cruz D'Cruz Consulting www.dcruz.net M: 07932 554993 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Ip phones
Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap You probably should do a little reading from the wiki and past postings as there is no lack of information on this topic. Cisco and Polycom phones rank the highest in terms of overall quality by those that have been exposed to lots of sip phones. Lots of sip phones in the middle, while the most inexpensive phones tend to be rated lower quality for many different reasons. When working with non-technical people and sip phones, they tend not to like Snom's and Grandstreams (and others) due to what technical people think are silly things. Those silly things are things like: - light weight phones that slide around the desk - displays that aren't readable unless you stand up - poor display images (including letters) - function keys that are not intuitive (or don't work as expected) - buttons that are hard to press - speaker phone functions that should never have been included since they don't work in a reasonable office environment - menues that are difficult to use by non-technical users, or are layered so deep it takes time to find commonly used functions - etc, etc, etc. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HooDaHek 0.6 Released
HooDaHek 0.6 has been released. So soon, you say? Well, the best laid plans of mice and men... Steven BerkHolz is a pretty sharp stick and said to me, Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call? Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of hoodahek_dbhandle anyway. SO: Version 0.6 has the following changes: - Got the correct version of hoodahek_dbhandle inserted, which has advanced error checking (yay) and also changes the CallerID in Asterisk if it performs a successful lookup in the HooDaHek database. Thanks to Steven BerkHolz for pointing out that rather obvious tidbit. As always, information and download linkage available here: http://www.nathanpralle.com/software/hoodahek.html Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HooDaHek 0.6 Released
Nathan Pralle wrote: HooDaHek 0.6 has been released. So soon, you say? Well, the best laid plans of mice and men... Steven BerkHolz is a pretty sharp stick and said to me, Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call? Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of hoodahek_dbhandle anyway. SO: Version 0.6 has the following changes: - Got the correct version of hoodahek_dbhandle inserted, which has advanced error checking (yay) and also changes the CallerID in Asterisk if it performs a successful lookup in the HooDaHek database. Thanks to Steven BerkHolz for pointing out that rather obvious tidbit. As always, information and download linkage available here: http://www.nathanpralle.com/software/hoodahek.html Nathan Does that mean I could use it with no instant messaging? I would like to have a local callerID database. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN vs NAT Helper
What is this sip-nat-helper thing, is there a website were we can get some info on it, partly thinking as the question before was relating to open source software, I would assume that I could download this thing. Dan On Wed, 14 Sep 2005 [EMAIL PROTECTED] wrote: If you have a linux box, then u can try sip-nat-helper for netfilter. Cheers. Mensaje citado por: Waldo Rubinstein [EMAIL PROTECTED]: I\'m wondering if anyone can recommend one over the other. I\'m mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional wealth of features it can add to the SIP services (e.g. voicemail, ivr, call queueing, etc). All of our clients are behind NATs, mainly basic NATs such as linksys routers behind DSL modems. I read on the wiki that STUN is not readily supported by most clients, so I don\'t know if its worth the effort or if we should just concentrate on getting SER working with Asterisk. Any ideas or suggestions? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Registrate desde http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y particip? de todos los beneficios del Portal Arnet. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [SPAM:***** SpamScore] Re: [SPAM:******** SpamScore] [Asterisk-Users] Call Transfer using SIP clients
I'm having the same problem you had Frank, so I'm pleased you came up with a fix. No luck for me yet! Incoming outgoing calls work fine using X-Lite, I just cannot transfer. It's the first time I've ventured in to features.conf so I'm likely doing something silly. I'd be grateful if you could have a look. I've posted (parts of) my sip and features .conf files below. Do I need something special in extensions.conf? What's supposed to happen when I dial *1, do I hear a special dialtone and then enter the extension? Thanks, Hugh ** features.conf (mostly default): ; Sample Parking configuration ; [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in ;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;findslot = next ; Continue to the 'next' parking space. Defaults to 'first' available ;pickupexten = *8 ; Configure the pickup extension. Default is *8 featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = *1 ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2; Attended transfer ** sip.conf : [301] ; My phone ; type=friend ; friend means this device takes and makes calls username=301 ; Username on device callerid=Me999-999- secret= ; Password for device nat=no host=dynamic ; This host is not on the same IP addr every time context=internal ; Inbound calls from this host go here mailbox=301 ; Activate the message waiting light if this ; voicemail box has messages in it canreinvite=no; Leave this alone for now; see archives for details disallow=all allow=gsm allow=ulaw allow=alaw On 7/5/05, Frank Schoep [EMAIL PROTECTED] wrote: On Tuesday 05 July 2005 09:29, Frank Schoep wrote: If I find out how to get it working, I will append that information to the thread so others can reuse that knowledge later on, I'm sure someone will appreciate it. So, I just got X-Lite working alongside Asterisk, the problem was (call it a premonition) the fact that I set them up to send DTMFs in band. Setting this option to disabled made the X-Lite softphone work flawlessly. I hope that helps someone. Sincerely, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BackgroundDetect problem
Hi all, I hate to ask such a simple question, but it has stumped me over the past couple of days. I have 2 asterisk servers connected to the house lan and also via a crossover ethernet cable. The original purpose of the crossover was to create a private lan for TDMoE. I have a TE410P in each machine using PRI. I also have setup SIP and IAX2 between both machines. What I'm doing is generating the call via the manager on the 1st machine and having it dial via Zap, IAX2, or SIP to the 2nd machine. On the 2nd machine, it plays vm-extension or my 'hello' file. My problem is that when using BackgroundDetect with SIP or IAX2, it does not work with an auto-generated call to the 2nd machine. It works fine using the PRI connection. It also works fine if I have it generate the call and then dial my SIP phone on the 2nd machine and I say hello. I tried monitoring the calls and I can hear the message being played when I just have it dial the 2nd machine and not dial the phone. It never goes to the talk extension when using SIP or IAX, but does when using Zap. I'm always using the defaults for BackgroundDetect and the same sound file when I call. I can't for the life of me understand why it will work using Zap but not IAX2 or SIP. It works if it dials my SIP phone on the 2nd machine. I tried recording that call and using that for the basis of the sound byte on the 2nd machine and it doesn't work for the automated calls. 1st machine (ZAP) - 2nd machine = OK 1st machine (IAX2) - 2nd machine = NO 1st machine (SIP) - 2nd machine = NO 1st machine (ZAP) - 2nd machine - SIP phone = OK 1st machine (IAX2) - 2nd machine - SIP phone = OK 1st machine (SIP) - 2nd machine - SIP phone = OK What's going on here? Thanks, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HooDaHek 0.6 Released
Paul wrote: Nathan Pralle wrote: HooDaHek 0.6 has been released. snip As always, information and download linkage available here: http://www.nathanpralle.com/software/hoodahek.html snip Does that mean I could use it with no instant messaging? I would like to have a local callerID database. Absolutely. Just don't run the hoodahek_notify script (don't copy it to your AGI folder, and don't go and put it in your extensions.conf). Everything else should work finethey're all modular. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HooDaHek 0.6 Released
Some websites allow you to look up a phone number and return a name/address. As a possible add-on to this, I have an agi script that looks up caller ID information on a few of these websites. It is written in C/C++. Currently these scripts are limited to Dutch numbers, since those are basically the only ones that I need. But I figure some kind of plug-in architecture could be made that allows looking up phone numbers from more than one source (currently just the hoodaheck mysql table, from what I understand). I am not sure if elsewhere in the world also websites like this exist (for other countries). I am posting this just like a (IMHO) useful idea. Any comments? Rene Kluwen Chimit -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nathan Pralle Sent: dinsdag 20 september 2005 17:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HooDaHek 0.6 Released HooDaHek 0.6 has been released. So soon, you say? Well, the best laid plans of mice and men... Steven BerkHolz is a pretty sharp stick and said to me, Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call? Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of hoodahek_dbhandle anyway. SO: Version 0.6 has the following changes: - Got the correct version of hoodahek_dbhandle inserted, which has advanced error checking (yay) and also changes the CallerID in Asterisk if it performs a successful lookup in the HooDaHek database. Thanks to Steven BerkHolz for pointing out that rather obvious tidbit. As always, information and download linkage available here: http://www.nathanpralle.com/software/hoodahek.html Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to distinguish the ringing and connected for zap channel
I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to connected even the callee is still ring. How could asterisk distinguish the ringing and connected in zap channel? thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ODBC Voicemail WEB Retrieval
Ok. I was sucessful in installing ODBC storage I'm using MySQL in the backend as it is. but all my drivers are now ODBC. I am running asterisk-cvs head as of last night 9/19/05 My question is this... the old voicemail.cgi script that allowed checking voicemail no longer works etc, and never did work for me without a static voicemail.conf file. Anyways.. that aside... how does one retrieve the longblob object from the database and present it to the user (upon authentication) via a website. I'd be happy to help someone with the www/php/mysql integration but I just dont know how to get blob's out and save to a temp file out of a database. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SMS using a PRI channel
Hi, On Wed, Sep 14, 2005 at 04:53:54PM +0200, Roger Schreiter wrote: I have some experience in sending SMSs using smsclient. I call the german Vodafone SMSC (01722278020), and smsclient takes approx 20 secs to send a SMS. The hardware is an Sedlbauer ISDN card. smsclient seems to be similar to yaps. http://www.smsclient.org http://freshmeat.net/projects/yaps/ Now, I want to do the same using asterisk and a digium PRI card. It is different. Asterisk sms application is designed for a protocol described in ETSI ES 201 912. (German: Festnetz SMS) I dialed using the manager with: action: originate channel: Zap/g4/01722278020 ... I assumed, the call will fail, because the remote end will become signalled a voice call, and imho the SMSC wouldn't answer a voice call, but expects data calls. It is a voice call and it has to be a voice call. FSK (Frequency Shift Keying) is used for modulation. Did anybody try sending SMS to german Vodafone or other SMSC mentioned in the smsclient package? Send the SMS using Telekom 0193010 gateway. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agent channel busy - how to stop it?
when a call file is used to place a call FROM an agent the agent is flagged as busy/unavail even if the call is subsequently transfered. call file has...Channel: AGENT/blah... Any way to stop the agent channel being flagged as busy? Cheers __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
I have used the pre20 package, with the latest CVS-head. COmpile goes cleanly, NO ERRORS. then I get this when I try to load asterisk -cvv [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handlerSep 20 14:00:23 WARNING[5924]: loader.c:554 load_modules: Loading module app_rxfax.so failed! No matter what I do it compiles clean but errors out with undefined symbol errors. Does anyone have a clue on this From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David SampsonSent: Thursday, September 15, 2005 12:17 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem I used the latest version (.3) and also the previous .2 ver (pre20). The spandsp seems to compile but when I download the rxfax/txfax .c files and drop them in the apps directory that is where I get the compile error. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ WeschkeSent: Wednesday, September 14, 2005 2:43 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem What version of spandsp are you attempting to compile in to the 1.0.9 tree? On 9/14/05, David Sampson [EMAIL PROTECTED] wrote: Anyone know how to fix this?gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HooDaHek 0.6 Released
Rene Kluwen wrote: Some websites allow you to look up a phone number and return a name/address. As a possible add-on to this, I have an agi script that looks up caller ID information on a few of these websites. It is written in C/C++. I'm not aware of websites like this in the USA or other countries, but that being said, I've not seen everything in the world, either. :) Sure, I'd be more than willing to do that sort of plugin to HooDaHek so you could prioritize lookups. In fact, I've been contemplating adding in a lot of functionality for doing things like interacting with cellphone APIs, etc. for various ways of notifying. Alternate data sources are something to consider, too. If anyone has more information on websites like this, I'd be happy to look into them. Thanks, Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax/TxFax - Compile Problem
On Tuesday 20 September 2005 15:10, Alexander Lopez wrote: I have used the pre20 package, with the latest CVS-head. COmpile goes cleanly, NO ERRORS. then I get this when I try to load asterisk -cvv [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Sep 20 14:00:23 WARNING[5924]: loader.c:554 load_modules: Loading module app_rxfax.so failed! No matter what I do it compiles clean but errors out with undefined symbol errors. Does anyone have a clue on this I had this problem too but I can remember exactly what triggered it or how I solved, I can say that with CVS HEAD and spands 0.0.2pre20 and the tx/rxfax that goes with it, everything worked. What version of spandsp are you using ? Did you install more than one version ? how did you installed them ? What version of the apps are you using ? are you sure you patched the Makefile correctly (and you are not using an old version, that happened to me) ? I have documented what I did to get fax working on http://www.voip-info.org/tiki-index.php?page=app_rxfax%20and%20app_txfax and most of the pages in the see also section. Be sure to check the comments, since I posted some important info there. Hope it helps. -- José Pablo Ezequiel Fernández ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HooDaHek 0.6 Released
Yellowpages.com has a reverse lookup on it. http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp As does whitepages: http://www.whitepages.com/10001/reverse_phone -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Pralle Sent: Tuesday, September 20, 2005 2:12 PM To: Rene Kluwen Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HooDaHek 0.6 Released Rene Kluwen wrote: Some websites allow you to look up a phone number and return a name/address. As a possible add-on to this, I have an agi script that looks up caller ID information on a few of these websites. It is written in C/C++. I'm not aware of websites like this in the USA or other countries, but that being said, I've not seen everything in the world, either. :) Sure, I'd be more than willing to do that sort of plugin to HooDaHek so you could prioritize lookups. In fact, I've been contemplating adding in a lot of functionality for doing things like interacting with cellphone APIs, etc. for various ways of notifying. Alternate data sources are something to consider, too. If anyone has more information on websites like this, I'd be happy to look into them. Thanks, Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax/TxFax - Compile Problem
Alexander Lopez wrote: I have used the pre20 package, with the latest CVS-head. COmpile goes cleanly, NO ERRORS. then I get this when I try to load asterisk -cvv [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Sep 20 14:00:23 WARNING[5924]: loader.c:554 load_modules: Loading module app_rxfax.so failed! No matter what I do it compiles clean but errors out with undefined symbol errors. Does anyone have a clue on this I'm having the same issues, so I've installed Asterisk on my laptop, did a fresh compile of libtiff and spandsp pre2-20 and started Asterisk. Asterisk app_txfax and app_rxfax compile without issues and Asterisk starts without complaining I'm going to remove and re-compile spandsp and libtiff tonight to see if it makes any difference with the effected machine or not. Doug -- Ben Franklin quote: Those who give up essential liberties for temporary safety deserve neither liberty nor safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of José Pablo Ezequiel Fernández Sent: Tuesday, September 20, 2005 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem On Tuesday 20 September 2005 15:10, Alexander Lopez wrote: I have used the pre20 package, with the latest CVS-head. COmpile goes cleanly, NO ERRORS. then I get this when I try to load asterisk -cvv [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Sep 20 14:00:23 WARNING[5924]: loader.c:554 load_modules: Loading module app_rxfax.so failed! No matter what I do it compiles clean but errors out with undefined symbol errors. Does anyone have a clue on this I had this problem too but I can remember exactly what triggered it or how I solved, I can say that with CVS HEAD and spands 0.0.2pre20 and the tx/rxfax that goes with it, everything worked. What version of spandsp are you using ? Did you install more than one version ? how did you installed them ? What version of the apps are you using ? are you sure you patched the Makefile correctly (and you are not using an old version, that happened to me) ? I have documented what I did to get fax working on http://www.voip-info.org/tiki-index.php?page=app_rxfax%20and%2 0app_txfax and most of the pages in the see also section. Be sure to check the comments, since I posted some important info there. Hope it helps. -- José Pablo Ezequiel Fernández ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users THANK YOU! Your Did you install more than one version ? Question did the trick!!! I had originaly installed version 3 of spandsp and for som unknown reasone it still had a symbolic link to it. Removed everything in /usr/lib/spandsp* and /usr/include/libspan* recomliled and installed and it works Somethimes it the little things we forget that get us!!! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
Try rm -rf /usr/include/spandsp* rm -rf /usr/lib/libspandsp* Then do a make install in the spandsp directory.. It may make you smile! It made me!! Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, September 20, 2005 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem Alexander Lopez wrote: I have used the pre20 package, with the latest CVS-head. COmpile goes cleanly, NO ERRORS. then I get this when I try to load asterisk -cvv [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Sep 20 14:00:23 WARNING[5924]: loader.c:554 load_modules: Loading module app_rxfax.so failed! No matter what I do it compiles clean but errors out with undefined symbol errors. Does anyone have a clue on this I'm having the same issues, so I've installed Asterisk on my laptop, did a fresh compile of libtiff and spandsp pre2-20 and started Asterisk. Asterisk app_txfax and app_rxfax compile without issues and Asterisk starts without complaining I'm going to remove and re-compile spandsp and libtiff tonight to see if it makes any difference with the effected machine or not. Doug -- Ben Franklin quote: Those who give up essential liberties for temporary safety deserve neither liberty nor safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax/TxFax - Compile Problem
What version of libtiff are you using. Has anyone tried 3.7.x with spandsp? Doug Lytle wrote: Alexander Lopez wrote: I have used the pre20 package, with the latest CVS-head. COmpile goes cleanly, NO ERRORS. then I get this when I try to load asterisk -cvv [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Sep 20 14:00:23 WARNING[5924]: loader.c:554 load_modules: Loading module app_rxfax.so failed! No matter what I do it compiles clean but errors out with undefined symbol errors. Does anyone have a clue on this I'm having the same issues, so I've installed Asterisk on my laptop, did a fresh compile of libtiff and spandsp pre2-20 and started Asterisk. Asterisk app_txfax and app_rxfax compile without issues and Asterisk starts without complaining I'm going to remove and re-compile spandsp and libtiff tonight to see if it makes any difference with the effected machine or not. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Ip phones
Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HooDaHek 0.6 Released
On Tue, 2005-09-20 at 14:31 -0400, Jonathan k. Creasy wrote: Yellowpages.com has a reverse lookup on it. http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp As does whitepages: http://www.whitepages.com/10001/reverse_phone http://directory.google.com/Top/Reference/Directories/Address_and_Phone_Numbers/ and lets not forget google itself (residential only aparently) phonebook:QUERY (smith, ca or 2025551212) There are a lot of them out there, used by stalkers every day. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: how to distinguish the ringing and connectedfor zap channel
Hi there, Basically, youare supposed to play arround with indications.conf To have the extensions configured with callprogress=yes but, be carefull because it is quite experimental. Also, what I did was to get an audio program (Cooledit, Adobe audition, or other), and you should use the spectral view (FFT Fast Fourier Transform), there you will be ableto see which frequencies the tones have, and their duration. Now, having said that I was half successfull in making it work, and I still have some problems, so if anybody else has a clear idea of what can be done, please shout in here ! :) Give a look to this: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+indications.conf Particullaryl the section: Generating a Tone Set And this: http://www.speech.kth.se/wavesurfer/ Cheers! Alchaemist Liu Peter [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to connected even the callee is still ring. How could asterisk distinguish the ringing and connected in zap channel? thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Canreinvite (yes, again)
use g711u for fax not 729 - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 19, 2005 4:21 PM Subject: [Asterisk-Users] T.38 Canreinvite (yes, again) I know this has been asked before, but I've checked the archives and I haven't found anybody that has given a definitive yes or no, just yeah, it should work.. If I have a T.38 gateway like a Cisco 5300 and a T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work? I have it setup and it doesn't work, so I want to know if I am doing something wrong, or if it just won't work. If I make a voice call, I see the media stream go from the gateway to the ata directly. When I fax, I see the stream go that way as well, but it is g.729. I see INVITE messages from my ATA that reference T.38, but they go to the * box, not the gateway and therefore * ignores it. Any thoughts? PA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ODBC Voicemail WEB Retrieval
On 9/20/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Ok. I was sucessful in installing ODBC storage I'm using MySQL in the backend as it is. but all my drivers are now ODBC. I am running asterisk-cvs head as of last night 9/19/05 My question is this... the old voicemail.cgi script that allowed checking voicemail no longer works etc, and never did work for me without a static voicemail.conf file. Anyways.. that aside... how does one retrieve the longblob object from the database and present it to the user (upon authentication) via a website. I'd be happy to help someone with the www/php/mysql integration but I just dont know how to get blob's out and save to a temp file out of a database. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I would like to add Asterisk Realtime Support to ARI (www.littlejohnconsulting.com). Please contact me off list. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax/TxFax - Compile Problem
Michael Welter wrote: What version of libtiff are you using. Has anyone tried 3.7.x with spandsp? I was running 3.7.2 without issues, but reverted to 3.5.7 because of issues I was trying to track down. Didn't do any better or worse then 3.5.7. Doug -- Ben Franklin quote: Those who give up essential liberties for temporary safety deserve neither liberty nor safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Ip phones
On 20:38, Tue 20 Sep 05, Florian Overkamp wrote: Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. I been using some Cisco phones for a while now. I started with converting them to SIP so they could connect to * Now with chan_sccp I reverted them all back to SCCP and they work awesome. Too bad they are so darn expensive, otherwise I wouldn't use anything else. Just my experience :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HooDaHek 0.6 Released
(trimmed) http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp http://www.whitepages.com/10001/reverse_phone http://directory.google.com/Top/Reference/Directories/Address_and_Phone_Numbers/ and lets not forget google itself (residential only aparently) phonebook:QUERY (smith, ca or 2025551212) Whitepages/Yellowpages (they use the same scripts, essentially) have done some impressive work at preventing scripts from getting information from their CGI. I have yet to work around it. The problem with these, of course, is that to integrate them into something useful you're either screen scraping or playing around with a lot of strange scripts. That's fine -- not like I haven't done that before, but for some you could easily violate their Terms of Service for doing so and get your IP banned or similar (esp. if your call volume is high). Although, the point might be that for residential * use, the volume might be low enough to warrant some lookups. I'll have to play with it a bit. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [ANNOUNCE] chan_capi-cm-0.6 released
Hi all, it took a while, but on sourceforge.net I added the new release 0.6 of chan_capi-cm driver. Note: dial string and capi.conf has changed. The main changes are: - added 'relaxdtmf'. - more BSD compatibility - correct PROGRESS handling - added verbose text for capi info/reason error messages. - use correct facility-selector for echo-cancel - added application capicommand() for CAPI based applications (removed standalone applications) - capicommand(RETRIEVE) can now be called from other channels - support ISDN hold (holdtype in capi.conf) - added HOLD/RETRIEVE for Asterisk indications. - added custom function VANITYNUMBER to convert letters into digits. - added CAPI Line Interconnect (native bridging) - use variable CONNECTEDNUMBER on Answer(). - set variable REDIRECTINGNUMBER on incomming call if it was diverted. - removed obsolete thread mutex - fixed dnid/exten/immediate handling on PtP. - receive a fax via CAPI is now done with capicommand(receivefax|...) and added stationid... - added config option 'immediate' to start pbx if no dnid has been received yet. - support 'type of number' (numbering-plan). - U-Law setting is now done in capi.conf instead of Makefile define. - on hangup, use hangupcause from other channel or from var PRI_CAUSE. - capi.conf structure changes: one own section for each interface, no global 'interfaces' any more. Section name will be interface name. - dial string changed: parameters like 'b' not as part of number any more. - send alert on alerting only (busy() and congestion() work now). - better overlap sending (new parameter 'o' for dialstring to send only the first two digits with CONNECT_REQ only, the remaining digits and even digits following the dial() command, will be send as INFO_REQ/Overlap). Have fun Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ODBC Voicemail WEB Retrieval
could you add it into cvs head? thanks.. 2005/9/20, Dan Littlejohn [EMAIL PROTECTED]: On 9/20/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Ok. I was sucessful in installing ODBC storage I'm using MySQL in the backend as it is. but all my drivers are now ODBC. I am running asterisk-cvs head as of last night 9/19/05 My question is this... the old voicemail.cgi script that allowed checking voicemail no longer works etc, and never did work for me without a static voicemail.conf file. Anyways.. that aside... how does one retrieve the longblob object from the database and present it to the user (upon authentication) via a website. I'd be happy to help someone with the www/php/mysql integration but I just dont know how to get blob's out and save to a temp file out of a database. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I would like to add Asterisk Realtime Support to ARI (www.littlejohnconsulting.com). Please contact me off list. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: how to distinguish the ringing and connectedfor zap channel
1) how to config callprogress=yes ? in extensions.conf? could you give me an example? 2) you means record the call (via zaptel) into a file and analyze it with audio tool? thanks.. 2005/9/20, Alchaemist [EMAIL PROTECTED]: Hi there, Basically, youare supposed to play arround with indications.conf To have the extensions configured with callprogress=yes but, be carefull because it is quite experimental. Also, what I did was to get an audio program (Cooledit, Adobe audition, or other), and you should use the spectral view (FFT Fast Fourier Transform), there you will be ableto see which frequencies the tones have, and their duration. Now, having said that I was half successfull in making it work, and I still have some problems, so if anybody else has a clear idea of what can be done, please shout in here ! :) Give a look to this: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+indications.conf Particullaryl the section: Generating a Tone Set And this: http://www.speech.kth.se/wavesurfer/ Cheers! Alchaemist Liu Peter [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to connected even the callee is still ring. How could asterisk distinguish the ringing and connected in zap channel? thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [SPAM:***** SpamScore] Re: [SPAM:******** SpamScore] [Asterisk-Users] Call Transfer using SIP clients
Figured it out. I didn't have tT in my dial command:
Dial(ZAP/1${ARG3},10,tT)
Thanks for posting your problem and solution. It sure helped me out...
Hugh
On 9/20/05, hugolivude [EMAIL PROTECTED] wrote:
I'm having the same problem you had Frank, so I'm pleased you came up
with a fix. No luck for me yet!
Incoming outgoing calls work fine using X-Lite, I just cannot transfer.
It's the first time I've ventured in to features.conf so I'm likely
doing something silly. I'd be grateful if you could have a look.
I've posted (parts of) my sip and features .conf files below. Do I
need something special in extensions.conf?
What's supposed to happen when I dial *1, do I hear a special dialtone
and then enter the extension?
Thanks,
Hugh
** features.conf (mostly default):
; Sample Parking configuration
;
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
;parkingtime = 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
transferdigittimeout = 3 ; Number of seconds to wait between digits
when transfering a call
courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
xfersound = beep; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot = next ; Continue to the 'next' parking space.
Defaults to
'first' available
;pickupexten = *8 ; Configure the pickup extension. Default is
*8
featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
blindxfer = *1 ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record
atxfer = *2; Attended transfer
** sip.conf :
[301]
; My phone
;
type=friend ; friend means this device takes and makes calls
username=301 ; Username on device
callerid=Me999-999-
secret= ; Password for device
nat=no
host=dynamic ; This host is not on the same IP addr every time
context=internal ; Inbound calls from this host go here
mailbox=301 ; Activate the message waiting light if this
; voicemail box has messages in it
canreinvite=no; Leave this alone for now; see archives for details
disallow=all
allow=gsm
allow=ulaw
allow=alaw
On 7/5/05, Frank Schoep [EMAIL PROTECTED] wrote:
On Tuesday 05 July 2005 09:29, Frank Schoep wrote:
If I find out how to get it working, I will append that information to the
thread so others can reuse that knowledge later on, I'm sure someone will
appreciate it.
So, I just got X-Lite working alongside Asterisk, the problem was (call it a
premonition) the fact that I set them up to send DTMFs in band. Setting this
option to disabled made the X-Lite softphone work flawlessly. I hope that
helps someone.
Sincerely,
Frank
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RE: [Asterisk-Users] Cisco Ip phones
Have you tested Aastra. Works great with * and reasoable pricing Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: den 20 september 2005 20:57 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco Ip phones On 20:38, Tue 20 Sep 05, Florian Overkamp wrote: Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. I been using some Cisco phones for a while now. I started with converting them to SIP so they could connect to * Now with chan_sccp I reverted them all back to SCCP and they work awesome. Too bad they are so darn expensive, otherwise I wouldn't use anything else. Just my experience :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri gateway
On 20 Sep 2005, at 12:12, Baris Simsek wrote:Status: Provisioned, In Alarm, Down, Active Call your provider and ask them what they see. I guess they haven't enabled it yet.Tim.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HooDaHek 0.6 Released
OK Great, I'll give it a shot. I did find this other option http://archives.free.net.ph/message/20050309.013714.2d1bf446.en.html , but I do not really want to imbed this info in the asterisk database if I can have it external. (note: this other option did work when tested) -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Nathan Pralle [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] HooDaHek 0.6 has been released. So soon, you say? Well, the best laid plans of mice and men... Steven BerkHolz is a pretty sharp stick and said to me, Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call? Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of hoodahek_dbhandle anyway. SO: Version 0.6 has the following changes: - Got the correct version of hoodahek_dbhandle inserted, which has advanced error checking (yay) and also changes the CallerID in Asterisk if it performs a successful lookup in the HooDaHek database. Thanks to Steven BerkHolz for pointing out that rather obvious tidbit. As always, information and download linkage available here: http://www.nathanpralle.com/software/hoodahek.html Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstriCon 2006 Location
Matt Fredrickson wrote: On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote: Senad J wrote: If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) I would definitively agree! Yes, but what would one do there? One who doesn't gamble, drink, or carouse, that is. I am making my first trip to LV later this Fall, and I dread it. I can't imagine what I'll be able to find to do when I'm not at the conference. It's ok, I don't either :-) I was actually kind of wondering the same thing. I'm sure there's something to do that doesn't involve all of that. Matt, In Las Vegas, I just might join you... You can pretty easily pay $12 for a drink in a club/bar! -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom-320 badly garbled audio
Hello, I just bought a Snom-320 from ATAComm. I plugged it into my LAN, registered it with *, etc. All my other SIP gear is Sipura and works fine, both on the LAN and over the Internet. The new Snom seems like it can't process the audio from the handset mic. A steady tone is garbled, even on the LAN. I've contacted ATAComm, Snom and the company representing Snom in the US. So far, ATAComm hasn't helped at all. The tech just said I dunno, and referred me to the US Snom rep co. Snom has replied, but the time zone differential and language barrier is making the process tedious. My * server is running Asterisk 1.0.9, zaptel 1.0.9.1, libpri 1.0.9. The frustrating part of this is that the Sipura gear works great. So, I have a hard time accepting that it's an * or LAN issue. Does anyone out there have Snom 320 phones in use? Are you experiencing garbled audio from the handset? Audio in works fine. But nobody can understand what I say back to them. I upgraded the Snom-320 to the latest firmware, v4.2, but that did not clear the problem. I've retooled so that I'm forcing ulaw, as I found that some folks have had bad luck with GSM. But I've tried both. Thanks Darren ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: how to distinguish the ringing and connectedfor zap channel
i checked the document about indicator.conf nd it is used to generator
the tone of busy, ringing, congestion or dialtone. Bt how can I detect
it in extension.conf?
I hope to know whether the callee is answered the call, or know the
duration of answered time. but even the callee doesnt picked the call,
the status was changed to ANSWER, and the ${ANSWEREDTIME} includes the
ring time for zaptel channel.
How can asterisk get the correct ${ANSWEREDTIME} for zaptel chanel?
thanks.
2005/9/20, Alchaemist [EMAIL PROTECTED]:
Hi there,
Basically, youare supposed to play arround with indications.conf
To have the extensions configured with callprogress=yes but, be
carefull because it is quite experimental.
Also, what I did was to get an audio program (Cooledit, Adobe
audition, or other), and you should use the spectral view (FFT Fast Fourier
Transform), there you will be ableto see which frequencies the tones have,
and their duration.
Now, having said that I was half successfull in making it work,
and I still have some problems, so if anybody else has a clear idea of what
can be done, please shout in here ! :)
Give a look to this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+indications.conf
Particullaryl the section: Generating a Tone Set
And this:
http://www.speech.kth.se/wavesurfer/
Cheers!
Alchaemist
Liu Peter [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
I have a TDM card in a asterisk machine.
I found that once I used it to call out, the call status changed to
connected even the callee is still ring.
How could asterisk distinguish the ringing and connected in zap channel?
thanks.
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Re: [Asterisk-Users] RxFax/TxFax - Compile Problem
On Tuesday 20 September 2005 15:36, Michael Welter wrote: What version of libtiff are you using. Has anyone tried 3.7.x with spandsp? My setup: tiff-3.7.3 * spandsp-0.0.2_pre20 * Asterisk HEAD with app_[rt]xfax-0.0.2_pre20 * These are Gentoo packages. It compiled, it started, it worked, sending and receiving faxes. I am now about to test with the Asterisk Gentoo packages, I'd expect it to work, otherwise I'll update the Gentoo packages (if anybody is interested on it). -- José Pablo Ezequiel Fernández ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: HooDaHek 0.6 Released
Steven wrote: I did find this other option http://archives.free.net.ph/message/20050309.013714.2d1bf446.en.html , but I do not really want to imbed this info in the asterisk database if I can have it external. (note: this other option did work when tested) Yeah, I tried that when I first started this project. Then I decided that I wanted something that had the ability to grown beyond that simple format, and an external AGI script setup seemed the best. I now have plans for the database to extend beyond simply holding numbers and names, but to holding addresses, contact information, and more importantly, to holding information about what Asterisk should DO with the call -- send it to voicemail, send it to a torture script, etc -- without having to write the dialplan for it. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax/TxFax - Compile Problem
This kind of mistakes are very common, I made them myself a couple of times, that's is why instead of going around removing and coping and symlinking files I prefeer to use the packages: emerge spandsp would do the trick. On Tuesday 20 September 2005 15:38, Alexander Lopez wrote: Try rm -rf /usr/include/spandsp* rm -rf /usr/lib/libspandsp* Then do a make install in the spandsp directory.. It may make you smile! It made me!! Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, September 20, 2005 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem Alexander Lopez wrote: I have used the pre20 package, with the latest CVS-head. COmpile goes cleanly, NO ERRORS. then I get this when I try to load asterisk -cvv [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Sep 20 14:00:23 WARNING[5924]: loader.c:554 load_modules: Loading module app_rxfax.so failed! No matter what I do it compiles clean but errors out with undefined symbol errors. Does anyone have a clue on this I'm having the same issues, so I've installed Asterisk on my laptop, did a fresh compile of libtiff and spandsp pre2-20 and started Asterisk. Asterisk app_txfax and app_rxfax compile without issues and Asterisk starts without complaining I'm going to remove and re-compile spandsp and libtiff tonight to see if it makes any difference with the effected machine or not. Doug -- Ben Franklin quote: Those who give up essential liberties for temporary safety deserve neither liberty nor safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- José Pablo Ezequiel Fernández ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Ip phones
On 21:30, Tue 20 Sep 05, Anders Svensson wrote: Have you tested Aastra. Works great with * and reasoable pricing Nope, haven't seen any phone of them in real life yet. Right now we deploy snom's for the price/quality rate they deliver. I find them very stable and nice phones. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk vertical service activation codes
Anybody know anything about using Asterisk vertical service activation codes as described in the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+vertical+service+activation+codes Specifically I'm interested in *0 that (apparently) flashes an external trunk on bridged channel. Nothing seems to happen when I use it though. Do I need to do something special in extensions.conf or perhaps features.conf to get this to work? Thanks, Hugh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HooDaHek 0.6 Released
I played around with finding the right place to call the agi.
Since my config started as [EMAIL PROTECTED], there are a lot of macros that
complicate things.
I put the agi in the macro-dial and it is working as expected. (just the
CLID record and change)
Thanks for the new tool.
ref:
[macro-dial]
exten = s,1,GotoIf($[ ${MACRO_CONTEXT} = macro-rg-group ]?4:2) ; if
this is from rg-group, don't strip prefix
exten =
s,2,agi,hoodahek_dbhandle|${CALLERIDNAME}|${CALLERIDNUM}|${UNIQUEID}
exten = s,3,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?4:4) ; check for ring-group prefix
exten = s,4,SetCIDName(${CALLERIDNAME:${LEN(${RGPREFIX})}}) ; strip off
prefix
exten = s,5,AGI,dialparties.agi
exten = s,6,NoOp(Returned from dialparties with no extensions to call)
exten = s,7,SetVar(DIALSTATUS=BUSY)
exten = s,10,Dial(${ds})
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
Steven [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
OK Great, I'll give it a shot.
I did find this other option
http://archives.free.net.ph/message/20050309.013714.2d1bf446.en.html , but
I do not really want to imbed this info in the asterisk database if I can
have it external. (note: this other option did work when tested)
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - --
- - --- - - -- - -- -- - --
Nathan Pralle [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
HooDaHek 0.6 has been released.
So soon, you say? Well, the best laid plans of mice and men...
Steven BerkHolz is a pretty sharp stick and said to me, Why don't you
have HooDaHek change the CallerID when it looks up the name in the
database on an incoming call? Much head smacketh ensued, and as I made
that change for Steven, I noticed that I had the way wrong version of
hoodahek_dbhandle anyway.
SO: Version 0.6 has the following changes:
- Got the correct version of hoodahek_dbhandle inserted, which has
advanced error checking (yay) and also changes the CallerID in Asterisk
if it performs a successful lookup in the HooDaHek database. Thanks to
Steven BerkHolz for pointing out that rather obvious tidbit.
As always, information and download linkage available here:
http://www.nathanpralle.com/software/hoodahek.html
Nathan
--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
-
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[Asterisk-Users] MySQL and Asterisk
Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ${DIALSTATUS} problems
I'm not expert on this matter,but base on experience that is a normal situation on SIP/IAX channels since the call made is answered by the other end regardless of the situation you might found.
I'm on PRI ISDN and for me dialstatus and hangupcause works pretty good.
Regards
Jb
On 9/15/05, Mark Edwards [EMAIL PROTECTED] wrote:
Hi.I'm dialling two numbers - one that's unobtainable, one that's busy.${DIALSTATUS} is coming back ANSWER each time right before the channels hang up.
Am using the following dialplan macro to dial out.[macro-advdial]exten = s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximumexten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL
,CONGESTION,ANSWER)exten = s-CHANUNAVAIL,1,NoOp(CHANUNAVAIL)exten = s-CHANUNAVAIL,2,UserEvent(ChannelUnavailable|Account: ${ACCOUNTCODE}^${CALLERIDNUM})exten = s-CONGESTION,1,NoOp(CONGESTION)
exten = s-CONGESTION,2,UserEvent(Congestion|Account: ${ACCOUNTCODE}^${CALLERIDNUM})exten = s-ANSWER,1,NoOp(ANSWER)exten = s-ANSWER,2,UserEvent(Answer|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-BUSY,1,NoOp(BUSY)exten = s-BUSY,2,UserEvent(Busy|Account: ${ACCOUNTCODE}^${CALLERIDNUM})exten = s-NOANSWER,1,NoOp(NOANSWER)exten = s-NOANSWER,2,UserEvent(NoAnswer|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answerOutbound calls are made using Manager originate interface from a meetme room channel Local/4000/n where 4000 is an extension which accesses the meetme room.
ITSP is terminating outbound calls to me via IAX2.I need to be able to see the CAUSE CODE status of the call if it is answered, CONGESTED or BUSY.my ITSP is in Australia - as am I.the IAX2 debug clearly indicates a zero CAUSE CODE on most call cases.
Any idea what I might be able to do to make the CAUSE CODE a little more meaningful?Also, does ${DIALSTATUS} or ${HANGUPCAUSE} work better on PRI?Cheers,Mark.
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[Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel, = Kind of solution...
Yes Darren. The problem is the same using Zap or SIP. I had no oportunity to verify that using IAX or E1/T1. Rgds, Ricardo Poppi. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom-320 badly garbled audio
Hello, On Tue, 20 Sep 2005, Darren Ellis wrote: Hello, I just bought a Snom-320 from ATAComm. I plugged it into my LAN, registered it with *, etc. All my other SIP gear is Sipura and works fine, both on the LAN and over the Internet. The new Snom seems like it can't process the audio from the handset mic. A steady tone is garbled, even on the LAN. I've contacted ATAComm, Snom and the company representing Snom in the US. So far, ATAComm hasn't helped at all. The tech just said I dunno, and referred me to the US Snom rep co. Snom has replied, but the time zone differential and language barrier is making the process tedious. My * server is running Asterisk 1.0.9, zaptel 1.0.9.1, libpri 1.0.9. The frustrating part of this is that the Sipura gear works great. So, I have a hard time accepting that it's an * or LAN issue. Does anyone out there have Snom 320 phones in use? Are you experiencing garbled audio from the handset? Audio in works fine. But nobody can understand what I say back to them. I upgraded the Snom-320 to the latest firmware, v4.2, but that did not clear the problem. I've retooled so that I'm forcing ulaw, as I found that some folks have had bad luck with GSM. But I've tried both. We've had these problems with Snom 190s in conjunction with el-cheapo 5-Port Switches. If you run the Snom in such an environment try using another (better) switch or if you can attach the el-cheapo switch to the PC-Port of the Snom. Regards Torsten Thanks Darren ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Media Online Internet Services Marketing GmbH Torsten Krueger [EMAIL PROTECTED] fon: 49-231-5575100fax: 49-231-55751098 Kurze Str. 10 D-44137 Dortmund ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Ip phones
We have tested this phone with a Asterisk system and deliver the phone with pre installed SIP-firmware without License What about the license?? And do you have to buy a license and changing the phone to sip protocol looks scary :( and time consuming with 100 phones not all suppliers will do it for you, and does any of you know a supplier in the netherlands with good pricing neonova is way too expensive -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Michiel van Baak Verzonden: dinsdag 20 september 2005 20:57 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [Asterisk-Users] Cisco Ip phones On 20:38, Tue 20 Sep 05, Florian Overkamp wrote: Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. I been using some Cisco phones for a while now. I started with converting them to SIP so they could connect to * Now with chan_sccp I reverted them all back to SCCP and they work awesome. Too bad they are so darn expensive, otherwise I wouldn't use anything else. Just my experience :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple PCI cards
Did you make any special configuration with the switch on the card? I have 2 TE400P that I haven't being able to use on 1 server. jb On 8/28/05, Asterisk [EMAIL PROTECTED] wrote: I have 2 TE410P's and a TDM400P in same machine without issuesBart-Original Message- From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ]On Behalf Of Damon EstepSent: Sunday, August 28, 2005 10:59 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Multiple PCI cards -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Garth van Sittert Sent: Sunday, August 28, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Multiple PCI cards Hi All Does anyone know if multiple Digium cards on a single machine will bea problem. Machine specs:Dual Zeon 3.0GHz on Intel server board. Cards: TE411P, TDM400P, TDM400P I will turn off all unnecessary PCI devices; USB, parallel, serial,etc... ThanksHave not tried it since November 2004, but at that time I ended upreplacing the FXO/FXS cards with sipura SPA3000 ?(check model number, its been awhile). Each one gave 1FXO/1FXS port so 4 of them replaced 2 4port TDM cads. Works well.Again, this was almost a year ago, so look for more feedback for usersthat have tried it with current hardware/firmware/software. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Ip phones
I have a snom 360 installed but the woman that is operating it complains about it all the time i looked at it and sometimes when sh transfers a phonecall it will just hang and stays in the phone the snom does not have connection to the line you can only see the line is still there in the display it tells you connected i think it's something like she don't push the buttons in good enough. But they complain about many things mostly they have to look inside there company phones are ringing but nobody answers them :) -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Michiel van Baak Verzonden: dinsdag 20 september 2005 22:01 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [Asterisk-Users] Cisco Ip phones On 21:30, Tue 20 Sep 05, Anders Svensson wrote: Have you tested Aastra. Works great with * and reasoable pricing Nope, haven't seen any phone of them in real life yet. Right now we deploy snom's for the price/quality rate they deliver. I find them very stable and nice phones. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HooDaHek 0.6 Released
On 9/20/05, Nathan Pralle [EMAIL PROTECTED] wrote: database on an incoming call? Much head smacketh ensued, and as I made Thou hast confused the present tense with the present participle. Thou couldest have written smacketh head smartly but perchance it is better to write there was much head-smacking and gnashing of teeth in this case, if thou so desirest to express thyself in the old tongue. The eth suffix is oft abused, and oft he who writeth it knoweth not the rules. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HooDaHek 0.6 Released
Thou hast confused the present tense with the present participle. Thou couldest have written smacketh head smartly but perchance it is better to write there was much head-smacking and gnashing of teeth in this case, if thou so desirest to express thyself in the old tongue. The eth suffix is oft abused, and oft he who writeth it knoweth not the rules. Quite right. My sincerest apologies to thee. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple PCI cards
On Tue, Sep 20, 2005 at 04:33:12PM -0400, Joan Bautista wrote: Did you make any special configuration with the switch on the card? I have 2 TE400P that I haven't being able to use on 1 server. IIRC, the T400Ps and E400Ps had a few problems with multiple cards together... Unless you're mistakenly meaning the TDM400Ps. If so, you should definitely be able to get a couple of those working in a box. -- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Differ between private and out of area?
usecallerid=yes hidecallerid=no callerid=asreceived usecallingpres=yes callwaiting=no callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes echotraining=yes group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=10 context=pstn rxgain=8.15 txgain=2.0 signalling=fxs_ks channel = 1 - Original Message - From: Rich Adamson [EMAIL PROTECTED] Paste the section from zapata.conf that handles the x101p. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: HooDaHek 0.6 Released
Steven,
Do you think the below dialplan would be typical for almost any
[EMAIL PROTECTED] setup? If so, I'll add it as supplimental documentation
for HooDaHek for those wanting to use it on [EMAIL PROTECTED]
Thanks,
Nathan
Steven wrote:
I played around with finding the right place to call the agi.
Since my config started as [EMAIL PROTECTED], there are a lot of macros that
complicate things.
I put the agi in the macro-dial and it is working as expected. (just the
CLID record and change)
Thanks for the new tool.
ref:
[macro-dial]
exten = s,1,GotoIf($[ ${MACRO_CONTEXT} = macro-rg-group ]?4:2) ; if
this is from rg-group, don't strip prefix
exten =
s,2,agi,hoodahek_dbhandle|${CALLERIDNAME}|${CALLERIDNUM}|${UNIQUEID}
exten = s,3,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?4:4) ; check for ring-group prefix
exten = s,4,SetCIDName(${CALLERIDNAME:${LEN(${RGPREFIX})}}) ; strip off
prefix
exten = s,5,AGI,dialparties.agi
exten = s,6,NoOp(Returned from dialparties with no extensions to call)
exten = s,7,SetVar(DIALSTATUS=BUSY)
exten = s,10,Dial(${ds})
--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
-
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Re: [Asterisk-Users] MySQL and Asterisk
Dan Journo wrote: Is there a guide anywhere which runs through how to set up asterisk with mysql? What, exactly, are you trying to do with MySQL and *? Access MySQL from the DialPlan: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MYSQL CDR record keeping in MySQL: http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysql RealTime Configuration: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime If all of these are confusing, feel free to ask on here what your specific questions are. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL and Asterisk
Dan Journo wrote: Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo What do you want to do with mysql? Did you read on the wiki? There is tons of info there. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Ip phones
On 22:28, Tue 20 Sep 05, Sander wrote: We have tested this phone with a Asterisk system and deliver the phone with pre installed SIP-firmware without License What about the license?? And do you have to buy a license and changing the phone to sip protocol looks scary :( and time consuming with 100 phones not all suppliers will do it for you, and does any of you know a supplier in the netherlands with good pricing neonova is way too expensive I got mine from www.centralpoint.nl As far as I know they only deliver the phones with SCCP image. But as you can read in my previous mail this is no problem, simply install chan_sccp. If you want the phones to run SIP, you have to buy a license for the SIP image. Centralpoint has them too. Changing the phones to SIP is really easy. Simply edit the lddefault.cfg so it will list the SIP image file. Put the SIP image and the lddefault.cfg file on your tftp server and every cisco rebooting will be converted to SIP. Reverting this process is the same (I just did it 3 weeks ago). Put the lddefault.cfg that states the SCCP image and the SCCP image on the tftp server and reboot the phones. I haven't tested the bigger cisco phones, but the 7905 has totally no trouble when converting from SCCP to SIP and viceversa. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom-320 badly garbled audio
You can always take a PCAP (Ethereal) trace from the phone's web page and analyze it with the RTP Statistics tool in Ethereal. That should give you a hint whats up with jitter Co. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Ellis Sent: Tuesday, September 20, 2005 10:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Snom-320 badly garbled audio Hello, I just bought a Snom-320 from ATAComm. I plugged it into my LAN, registered it with *, etc. All my other SIP gear is Sipura and works fine, both on the LAN and over the Internet. The new Snom seems like it can't process the audio from the handset mic. A steady tone is garbled, even on the LAN. I've contacted ATAComm, Snom and the company representing Snom in the US. So far, ATAComm hasn't helped at all. The tech just said I dunno, and referred me to the US Snom rep co. Snom has replied, but the time zone differential and language barrier is making the process tedious. My * server is running Asterisk 1.0.9, zaptel 1.0.9.1, libpri 1.0.9. The frustrating part of this is that the Sipura gear works great. So, I have a hard time accepting that it's an * or LAN issue. Does anyone out there have Snom 320 phones in use? Are you experiencing garbled audio from the handset? Audio in works fine. But nobody can understand what I say back to them. I upgraded the Snom-320 to the latest firmware, v4.2, but that did not clear the problem. I've retooled so that I'm forcing ulaw, as I found that some folks have had bad luck with GSM. But I've tried both. Thanks Darren ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fixlocalprefix error
Anyone know why I would be getting this error? All calls go through without problem but I get the following message: fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL and Asterisk
Ive already set up the cdr mysql. Now im trying to add realtime now but stuck on how to do it. those links didnt really help much. and the cli doesnt provide much info on what is going on. any help would be appreciated. Thanks Dan On 9/20/05, Nathan Pralle [EMAIL PROTECTED] wrote: Dan Journo wrote: Is there a guide anywhere which runs through how to set up asterisk with mysql? What, exactly, are you trying to do with MySQL and *?Access MySQL from the DialPlan:http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MYSQL CDR record keeping in MySQL:http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysqlRealTime Configuration: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTimeIf all of these are confusing, feel free to ask on here what yourspecific questions are.Nathan--- Nathan E. PralleGive the Director a Serpent Deflectorwww.nathanpralle.com- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Hardware Interrupts; Who is it?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, September 19, 2005 4:52 PM To: Asterisk Users Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it? CPU0 CPU1 0: 85 1703809954IO-APIC-edge timer 8: 0 0IO-APIC-edge rtc 9: 0 1 IO-APIC-level acpi 14: 0 31IO-APIC-edge ide0 177: 0 17840313 IO-APIC-level megaraid 185: 0 1817423967 IO-APIC-level eth0 193: 0 40198530 IO-APIC-level eth1 201: 0 3507106255 IO-APIC-level wanpipe1, wanpipe2, wanpipe3, wanpipe4 NMI: 0 0 LOC: 1633394197 1633394188 ERR: 0 MIS: 0 Might be your Sangoma board using up the cpu...? Other likely candidate is that its your so-called-hardware RAID (megaraid). They call it hardware, but its really software raid because the raid code runs on your main processor. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ${DIALSTATUS} problems
I met same problem when dial via zap channel.
Does anyone know how to solve it?
thanks.
2005/9/15, Mark Edwards [EMAIL PROTECTED]:
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang
up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten = s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL
,CONGESTION,ANSWER)
exten = s-CHANUNAVAIL,1,NoOp(CHANUNAVAIL)
exten =
s-CHANUNAVAIL,2,UserEvent(ChannelUnavailable|Account:
${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-CONGESTION,1,NoOp(CONGESTION)
exten = s-CONGESTION,2,UserEvent(Congestion|Account:
${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-ANSWER,1,NoOp(ANSWER)
exten = s-ANSWER,2,UserEvent(Answer|Account:
${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-BUSY,1,NoOp(BUSY)
exten = s-BUSY,2,UserEvent(Busy|Account:
${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-NOANSWER,1,NoOp(NOANSWER)
exten = s-NOANSWER,2,UserEvent(NoAnswer|Account:
${ACCOUNTCODE}^${CALLERIDNUM})
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
Outbound calls are made using Manager originate interface from a meetme room
channel Local/4000/n where 4000 is an extension which accesses the meetme
room.
ITSP is terminating outbound calls to me via IAX2.
I need to be able to see the CAUSE CODE status of the call if it is
answered, CONGESTED or BUSY.
my ITSP is in Australia - as am I.
the IAX2 debug clearly indicates a zero CAUSE CODE on most call cases.
Any idea what I might be able to do to make the CAUSE CODE a little more
meaningful?
Also, does ${DIALSTATUS} or ${HANGUPCAUSE} work better on PRI?
Cheers,
Mark.
--
regards,
Mark P. Edwards
FWD: 667917
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