[asterisk-users] CSTA support for asterisk
Hi, Can anybody tell me that is their CSTA support for asterisk sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] french promt
On Thu, 2006-07-27 at 11:23 +0200, Olivier Saulnier wrote: Follow this link: http://svn.digium.com/view/asterisk/sounds/fr/trunk/?rev=34575 You don't need to get them out of subversion. They are available in tarballs on the ftp. ftp://ftp.digium.com/pub/telephony/sounds When installing Asterisk 1.4, you will have a console menu system that you can use, which includes the ability to pick which sound packages you want based on language and format, and they will be automatically downloaded and installed. -- Russell Bryant Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting no Audio with G729
On Jul 27, 2006, at 2:06 PM, Wasif wrote: Hello, Recently I purchased g729 codec and installed in Tribox 1.1 (upgraded 1.1.1)/ Asterisk. I have pointed a DID from my carrier via SIP through g729 to asterisk. Problem is I am not getting any audio even though I am getting DTMF in asterisk. I am trying to run A2billing with asterisks. Configuration of carrier is asterisk is: [abc] allow=g729 context=c-DID dtmfmode=auto host=xxx.xxx.xxx.xxx insecure=very sendrpid=yes type=friend echo=no Make sure that the port 1-2 are open at your firewall? That is where the audio is being transmitted during an SIP call (usually). Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bugs.digium.com
On Thu, 27 Jul 2006, Russell Bryant wrote: Fine... so how do you file an enhancement request then? If there's no way to file an enhancement request, then this is the most appropriate place to file this. The bug tracker is really is not a good place for feature requests. You have to understand that this is the tool we have for managing all of Asterisk development contributions and bugs. If every user posted every feature they think should be implemented on there, it would make it much more difficult for us to manage. The developers *do* monitor the mailing lists. One of the major reasons we monitor the lists is to understand the issues that users are facing. Believe me, if you start a discussion on this mailing list regarding this issue, it will get noted, and if we are able, we will make improvements to make things easier for you. Also keep in mind that there are many thousands of users with many different ideas and drastically fewer developers that can implement them. That reminds me of other feature-requests I wanted to follow. So I would like to ask what the correct way would be for feature-requests? As far as I was told, the developers-list should be used. But what if no one is responding on the feature-request? How should we 'track' that request, see its status? Every open-source software development I know, uses the same tool for bugs and requests. Why else would the bug-tracker then have a 'feature-request' category? Maybe not *every* user should add feature-requests to the tracker and should start a discussion on developers-list first, but then, when decided to actually confirm this request, it can be added to the tracker ??? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk with CSTA using VAIL SIP TIM
Hi, Can anybody tell me how to configure asterisk for VAIL SIP TIM so that CSTA is utilized... sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Authorization
Sorry to dredge up an old topic, but could someone help me with this?
I need to accept and forward a call from a range of ip addresses
without any other authentication. (from-internal)
Does anyone have a small snipped of extensions.conf and sip.conf that I
can use to implement this?
Thanks in Advance
Aaron
Alexander Lopez wrote:
Exten = 123,1,NoOp(${SIPCHANINFO(recvip)})
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Sam Tam
Sent: Friday, February 10, 2006 3:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IP Authorization
Ah that is from the CLI but still unclear about how to setup
the extension.conf or etc..
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Alexander Lopez
Sent: Friday, February 10, 2006 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] IP Authorization
You can use the following:
switch3*CLI show function SIPCHANINFO
switch3*CLI
-= Info about function 'SIPCHANINFO' =-
[Syntax]
SIPCHANINFO(item)
[Synopsis]
Gets the specified SIP parameter from the current channel
[Description]
Valid items are:
- peeripThe IP address of the peer.
- recvipThe source IP address of the peer.
- from The URI from the From: header.
- uri The URI from the Contact: header.
- useragent The useragent.
- peername The name of the peer.
All the info you need is there.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf
Of Sam Tam
Sent: Thursday, February 09, 2006 9:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IP Authorization
Can you be more detail about the setup?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf
Of Olle E
Johansson
Sent: Friday, February 10, 2006 4:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP Authorization
Sam Tam wrote:
I think this is a question that has been discussed before.
But you see nowadays most carriers will provide thing like
SIP using
IP authorization rather than username and password and I am now
wondering whether Asterisk can do something like that or not?
In the voip channels as well as in manager you can set ACLs for the
connections you define.
/O
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[asterisk-users] Multiple Outbound SIP Trunks
I have 3 sip trunks registered with an outside provider, however asterisk always seems to work when going out the third trunk. Any way to round-robin this so that we can make more than one outbound call at a time? Thanks in advance, Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SNOM 360
On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fritz!Box Fon ATA
Hi, I have bought a Fritz!Box Fon ATA in eBay. Im trying to find information about configuration this box in Asterisk. Its possible use this box like a normal ATA (sipura 3000 ) receiving and making ISDN calls from Asterisk? Somebody has information in English about this box? Some example settings? Another problem is that firmware is in German. I have tried to change it but was not possible to use a difference language. Some ideas? Any help would be greatly appreciated Manuel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
Light the fiber to get to that 2000ft mark, then use directional antennas to cover the last 2000 ft via wireless. If necessary, you could even use unlicensed 900 mhz gear that runs 802.11g speeds (search for the Ubiquiti SR9), http://www.wlanparts.com/c=*/product/SR9 has it for $149 when in stock. Your best bet may be to try wireless the whole way and then try using the fiber if wireless isn't going to cut it. If you end up needing repeater stations, remember to factor in solar panel and battery costs sufficient to last the maximum number of sunless or minimal sun days. On Jul 27 2006, Manrique Feoli wrote: another thought, if you are in a bowl, all you need to find is line of sight to one common place from both ends, and place a repeater there. (you could also set two or three steps repeating the signal within points which have line of sight). I'm not sure but I think one repeater would be much cheaper than 20.000ft of copper + extenders + poles+ maintenance, lighning... (even thought you are in Copper Mountain !!, BTW nice spot ). if in the end you decide to go with ethernet, just beware of lighning!!! Brian Vincent (C) escribió: I know.. I know fiber would be ideal. We have single-mode all over the place. We even have some dark, unterminated strands within 2000ft of this location it makes me want to cry. Unfortunately lighting it up isnt an option we wouldnt gain anything because we couldnt connect to anything else to get us the last stretch. Trenching 2000ft isnt an option this is National Forest land and were not allowed to do that. As far as wireless no line of sight. This location sits in a little bowl at 11,200. So what Im left with is a 400pr, 22awg out to 3000. Then we jump on 200pr, 24awg aerial cable strung on the 3^rd longest high-speed quad chairlift (10,800 run). The last leg involves a short underground to another high-speed quad and down 6000. We can stick a powered repeater in the motor room of the first lift (so I guess a bit further than the original 12,000 I was thinking.) Yes, we do strange things. If youre really curious, heres a map of the campus environment we maintain: http://www.skireport.com/colorado/copper/trailmap/ --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bruce Reeves *Sent:* Thursday, July 27, 2006 4:03 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] long distance ethernet Asterisk I would really look towards fiber, the bandwidth and distance can easily be handled. On 7/27/06, *Manrique Feoli* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, *Brian Vincent (C)* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Two questions: 1. We need to run Ethernet out to a really long distance 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 10Mbps. 2. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. __ ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreePBX Inbound Route
Hi, I have SIP trunk. And I also have a lot of SIP clients. If I want to call from SIP trunk to the Asterisk SIP client, I need to create Inbound route for each endpoint. Maybe is possible to create an endpoint group, because I have a lot of SIP endpoints, and it takes a lot of time to create inbound routes. Or maybe it's only one way to do that. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rate engine AGI?
Is there an AGI out there which we can call from extensions.conf which will lookup a rate in a MySQL db based on the number the callerer dialed? We don't want anything with tons of features as we are doing all our coding, we just want something that will give us the rate and maybe permission to call or not call that country. We don't want it to bill for us or anything else because we have all that worked out, just need to get the rate. Any help would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom_acd_functions SIP trouble
At first, but if you checkout the version 1221, someone has fixed it. svn checkout http://svn.digium.com/svn/zaptel/trunk/ zaptel-trunk -r 1221 Regards, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Miller Sent: 28 July 2006 04:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom_acd_functions SIP trouble Did you by chance have to make changes to get Zaptel to compile? Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: Monday, July 24, 2006 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom_acd_functions SIP trouble I've got the same problem, the only version I can find that works is 30432, but the meetme conference does not compile in this version. A fix for the newest version for username/auth name would be great! - Original Message - From: James Fromm [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, July 24, 2006 7:24 PM Subject: [asterisk-users] Polycom_acd_functions SIP trouble I'm trying to use the latest revision of Bweschke's branch from SVN for polycom_acd_functions. Asterisk builds and runs without error but all SIP devices can't register when specifying a secret in sip.conf. The Polycom 601 I'm testing with and a copy of SJphone will not register. IAX from Idefisk works without error. The error all SIP devices get is: Jul 24 10:26:48 NOTICE[31524]: chan_sip.c:14203 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.95' - Username/auth name mismatch Commenting the definition of a secret in sip.conf for the device solves this. Here's the config for one of the devices. [1003] type=friend canreinvite=no host=dynamic username=1003 ; secret=stuff context=outbound callerid=Jimmy 1003 [EMAIL PROTECTED] nat=no Why won't this revision accept the definition of a secret? Am I missing something simple (stupid)? Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fritz!Box Fon ATA
Hi Manuel, I do not know the ATA, but we have a Fritz!Box fon, wich maybe is equal in setting it up. If you have any problems understanding the german setup, you can contact me, so I can help you in translating the needed Words :-) Normally you only have to do this on the Webinterface: Telefonie Internettelefonie Internetrufnummern Neue Internetrufnummer Internetrufnummer: Your VOIP number, or if using with isdn, then the msn. Do not use Internetrufnummer zum Anmelden verwenden! Registrar: the ip or host of your provider or Asterisk. If you have a own Asterisk use yur ip adress. There is a bug using hostnames. Benutzername: Username Passwort / Kennwort : password Do only fill out this fields, then it should work. If you put in any proxy or Stun Servers it may not work. (our experience) hth, Martin - Original Message - From: Manuel Dominguez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, July 28, 2006 9:25 AM Subject: [asterisk-users] Fritz!Box Fon ATA Hi, I have bought a Fritz!Box Fon ATA in eBay. I'm trying to find information about configuration this box in Asterisk. Its possible use this box like a normal ATA (sipura 3000.) receiving and making ISDN calls from Asterisk? Somebody has information in English about this box? Some example settings? Another problem is that firmware is in German. I have tried to change it but was not possible to use a difference language. Some ideas? Any help would be greatly appreciated Manuel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
On 7/28/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote: On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. As I understand it, you can set a QoS priority if the phone is in a VLAN. When you configure the (Tagged) VLAN, you can specify the priority of the packets in the VLAN. Otherwise, newer firmware allows the setting of TOS values IIRC. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PAP2T always busy on incoming calls with zaptel
Hi, I'm starting to use the new PAP2T instead of the old PAP2-NA for my new installations. I'm having a weird problem: when a call is comming from a zaptel channel (from a bri with bristuff driver) the PAP2T say BUSY to the SIP channel. I have disabled all the features like DND and call forward. If it's the last line for this number in the dialplan I can answer the call normally, but I can't use voicemail, because it jump to it each time. I have installed about 50 PAP-NA and never had this kind of problem. If the call is coming from an other PAP2T (via asterisk with canreinvite=no), everything is fine. This occur with asterisk 1.0.10 and 1.2.9.1 the firmware version for the PAP2T is 3.1.9(LSc) I am using a dialplan coming from another customer with a similar setup, but with PAP2-NA, where it's working fine. What can I do to fix this. Regards, Olivier ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip phone settings set when user registers
2006/7/27, Nik Engel [EMAIL PROTECTED]: User logs into any phone and the settings of the phone are always thesame. Meaning individual keyassignement is always the same.Hi,Do you mean :1. Without user logins, phones are unusable ? Or do you plan to offer default services (local calls for instance) for unidentifed users ? I'm not sure many phones offer special keys for login-logout. 2. What should happen when users change phones settings ? Shall these changes be saved somehow (during logoffs ?) and somewhere for latter reuse ? That implies phone config should be portable from one phone to another. That doesn't seem easy if phones are installed in different locations. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: alcatel ip touch 4068 ... sip?
2006/7/28, Leo Ann Boon [EMAIL PROTECTED]: (AstATN) wrote:Hi Cesc,Alcatel does not offer any SIP Terminal, it phone's are H.323 it's handlefor their own features usages. ( like ADSI type )Common misconception. Their phones are not H.323 despite claims in theirdocumentation. The server has to do the signaling conversion. The nativeprotocol is UAIP (User Agent IP) which runs over UDP.Hi,I've never heard of that (UAIP) before ! Do you have anything describing this protocol ?Would it be difficult to implement it inside Asterisk just like UNISTIM has been ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk+ooh323.. one way audio issue
Hi guys, I tried to make call from SIP channel to H323 using asterisk+ooh323. The SIP client is x-lite.The problem is that there is one way audio. I hear everything from h323 endpoint, and I see the messages also: Got RTP packet from 66.135.35.xx:5002 (type 3, seq 36250, ts 74400, len 33)Sent RTP packet to 212.183.41.xx:45956 (type 18, seq 22288, ts 70880, len 20) But the problem, when I talk via X-lite, or send dtmf tones, no audio is transfered, no RTP packets on asterisk console. My ooh323.conf: [general]port=1720bindaddr=0.0.0.0gateway=noh323id=ObjSysAsteriske164=100callerid=asteriskgatekeeper = DISABLEdisallow=allallow=g729allow=gsmallow=ulaw Note that I tried all combinations of faststart and h245tunneling, but no luck.Also tried with gsm and g729 codecs (that time X-pro was used) but same oneway audio.Asterisk version is 1.2.7.1 With full debug this is what I see in asterisk console: Jul 28 10:40:22 DEBUG[15775]: pbx.c:1674 pbx_extension_helper: Launching 'Dial'Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable STACK-test-381637790067-2.Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable STACK-test-381637790067-1.Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPCALLID.Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT.Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPURI.Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel OOH323/66.135.33.xx-14b0 to read format gsmJul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel SIP/66-9cbb to write format gsmJul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel SIP/66-9cbb to read format gsmJul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel OOH323/66.135.33.xx-14b0 to write format gsmJul 28 10:40:22 DEBUG[16193]: res_config_mysql.c:636 mysql_reconnect: MySQL RealTime: Everything is fine.Jul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set channel SIP/66-9cbb to read format gsmJul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set channel OOH323/66.135.33.xx-14b0 to write format gsmJul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set channel OOH323/66.135.33.xx-14b0 to read format gsmJul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set channel SIP/66-9cbb to write format gsmJul 28 10:40:23 DEBUG[15775]: chan_sip.c:2527 sip_answer: sip_answer(SIP/66-9cbb)Jul 28 10:40:23 DEBUG[15775]: channel.c:1956 ast_read: Dropping duplicate answer!Jul 28 10:40:23 DEBUG[16193]: res_config_mysql.c:125 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '66'Jul 28 10:40:23 DEBUG[16193]: res_config_mysql.c:636 mysql_reconnect: MySQL RealTime: Everything is fine.Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 6305: Match FoundJul 28 10:40:23 DEBUG[16207]: chan_sip.c:9442 check_pendings: Sending pending reinvite on '[EMAIL PROTECTED]'Jul 28 10:40:23 DEBUG[15775]: rtp.c:410 ast_rtcp_read: Got RTCP report of 84 bytesGot RTP packet from 87.116.143.xx:8000 (type 3, seq 1, ts 5920, len 33)Jul 28 10:40:23 DEBUG[15775]: rtp.c:1341 ast_rtp_write: Ooh, format changed from unknown to gsmSent RTP packet to 66.135.33.xx:5004 (type 3, seq 54271, ts 0, len 33)Got RTP packet from 87.116.143.xx:8000 (type 3, seq 2, ts 6080, len 33)Sent RTP packet to 66.135.33.xx:5004 (type 3, seq 54272, ts 160, len 33)Got RTP packet from 87.116.143.xx:8000 (type 3, seq 3, ts 6240, len 33)Sent RTP packet to 66.135.33.xx:5004 (type 3, seq 54273, ts 320, len 33)Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:1447 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundJul 28 10:40:23 DEBUG[16207]: chan_sip.c:1372 __sip_ack: Acked pending invite 102Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match FoundJul 28 10:40:23 DEBUG[16207]: chan_sip.c:6047 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060Got RTP packet from 66.135.33.xx:5004 (type 3, seq 55961, ts 3520, len 33)Jul 28 10:40:24 DEBUG[15775]: src/chan_h323.c:3045 ooh323_rtp_read: Oooh, format changed to 2Jul 28 10:40:24 DEBUG[15775]: channel.c:2348 set_format: Set channel OOH323/66.135.33.xx-14b0 to read format gsmJul 28 10:40:24 DEBUG[15775]: channel.c:2348 set_format: Set channel OOH323/66.135.33.xx-14b0 to write format gsmJul 28 10:40:24 DEBUG[15775]: rtp.c:1341 ast_rtp_write: Ooh, format changed from unknown to gsmSent RTP packet to 87.116.143.xx:8000 (type 3, seq
[asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
Hi,
We're seeing a problem on Asterisk 1.2.10 where when we get in in the
morning it's continually rotating the logs over and over again,
generating 100's of thousands of log rotated 0 byte files:-
/var/logs/asterisk # find . -type f -maxdepth 1 | wc -l
176930
/var/log/asterisk # find . -type f -maxdepth 1 -size 0 -exec mv {} nulls/ \;
/var/log/asterisk # find . -type f -maxdepth 1 | wc -l
69169
A segment of the relevant log is:-
Jul 25 06:33:42 VERBOSE[9638] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9638] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[9635] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9641] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9641] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[9638] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9644] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9644] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9641] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[9647] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9647] logger.c: Asterisk Queue Logger restarted
etc...
Has anyone else seen this or have any ideas what the problem may be?
Thanks,
--
Kenny Millington
Systems Developer
3aIT Limited
T: 0870 881 5097
F: 01403 248 105
E: [EMAIL PROTECTED]
W: http://www.3ait.co.uk
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[asterisk-users] Sending email after voicemail
Hi, I'm having trouble getting asterisk to send a voicemail message via email. I can do a mail [EMAIL PROTECTED] from the linux command line and I receive the email fine, and if I look in the exim4 logs it looks ok, has from user, to user and completed but no email is received. Any thoughts? Thanks, Dean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
check your cron jobs.
mayby there is asterisk -rx logger rotate executing too often ?
Hi,
We're seeing a problem on Asterisk 1.2.10 where when we get in in the
morning it's continually rotating the logs over and over again,
generating 100's of thousands of log rotated 0 byte files:-
/var/logs/asterisk # find . -type f -maxdepth 1 | wc -l
176930
/var/log/asterisk # find . -type f -maxdepth 1 -size 0 -exec mv {} nulls/ \;
/var/log/asterisk # find . -type f -maxdepth 1 | wc -l
69169
A segment of the relevant log is:-
Jul 25 06:33:42 VERBOSE[9638] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9638] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[9635] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9641] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9641] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[9638] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9644] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9644] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9641] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[9647] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9647] logger.c: Asterisk Queue Logger restarted
etc...
Has anyone else seen this or have any ideas what the problem may be?
Thanks,
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Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
Filip Drągowski wrote: check your cron jobs. mayby there is asterisk -rx logger rotate executing too often ? Nope - nothing in crontab. Hi, We're seeing a problem on Asterisk 1.2.10 where when we get in in the morning it's continually rotating the logs over and over again, generating 100's of thousands of log rotated 0 byte files:- snip -- Kenny Millington Systems Developer 3aIT Limited T: 0870 881 5097 F: 01403 248 105 E: [EMAIL PROTECTED] W: http://www.3ait.co.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR IP Authorization
Dear
This function retrieves the ip address of
the caller ,I want to import the value of (recvip) in the mysql cdr ,how can I
do that
exten = s,1,NoOp(${SIPCHANINFO(recvip)})
Regards
*
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Re: [asterisk-users] SIP client with video???
Hi, i have xlite too and it works without any problems. ps: what about ekiga? (www.ekiga.org) rich --- Joao Pereira [EMAIL PROTECTED] wrote: Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software. Does someone knows about a better software? Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer call in SIP
Hello, I am running TrixBox. if already in a call session from ZAPTEL to SIP, the user want to transfer the call to a different extension. The user have to dial *extension ? Any configuration is needed to be done in trixbox? Thanks Victor ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager interface
On 27 Jul 2006, at 22:42, Tielin Xu wrote: There are many ways to do the screen pop, I'd like to do this way: 1. Build the manager interface as an event server, which collect agent connet events. 2. Build a Java applet with the constant connection to the event server, each agent starts the Java applet at first task of each day 3. The event server sends the connect info to the computer which the agent registed, 4. The applet launch (pop up) the web based CRM application on agent computer with the caller's information 5. Agent terminates the CRM application when the call is termianted. Sure, that is pretty close to what we do, except that we don't use an event server. In our case the Java applet is a softphone that speaks IAX directly to asterisk. In our dial plan we have rules such that asterisk dials both the agent's hard phone (If they have one) and their copy of the applet. If you are interested, I'm sure I could arrange for you to have an eval copy of Corraleta (which is what we call the softphone applet). Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
As far as g.SHDSL is concerned, I think you are limited to 4mbit/s We have an internet connection at work is delivered over g.SHDSL, (at 1Mb/s) to a cisco 828 and if I remember right, the ping time is of the order of 10ms. Certainly no problem for SIP. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
on one cosnole a do asterisk -r on other i do asterisk -rx "logger rotate" and the result is -- Remote UNIX connection Asterisk Event Logger restarted Asterisk Queue Logger restarted -- Remote UNIX connection disconnected how often new log files are created ? = how many log files are created in 1 second ? there is some kind of regularity or it is done randomly (10logs/1s and another time 20logs/1s) ? for me it's looks like something causing regullary asterisk -rx "logger rotate" Filip Drągowski wrote: check your cron jobs. mayby there is asterisk -rx "logger rotate" executing too often ? Nope - nothing in crontab. Hi, We're seeing a problem on Asterisk 1.2.10 where when we get in in the morning it's continually rotating the logs over and over again, generating 100's of thousands of log rotated 0 byte files:- snip ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Ringing timer
Ok.Thanks a lot! I will try! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Canreinvite
How can I check if SIP re-invite is really working ? I'm trying it with two grandstream gxp2000. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Fritz!Box Fon ATA
-- Message: 11 Date: Fri, 28 Jul 2006 10:30:50 +0200 From: Olivier [EMAIL PROTECTED] Subject: Re: [asterisk-users] Sip phone settings set when user registers To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 2006/7/27, Nik Engel [EMAIL PROTECTED]: User logs into any phone and the settings of the phone are always the same. Meaning individual key assignement is always the same. Hi, Do you mean : 1. Without user logins, phones are unusable ? Or do you plan to offer default services (local calls for instance) for unidentifed users ? I'm not sure many phones offer special keys for login-logout. 2. What should happen when users change phones settings ? Shall these changes be saved somehow (during logoffs ?) and somewhere for latter reuse ? That implies phone config should be portable from one phone to another. That doesn't seem easy if phones are installed in different locations. Regards -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/b8a1c9 87/attachment-0001.htm -- Message: 12 Date: Fri, 28 Jul 2006 10:52:20 +0200 From: Olivier [EMAIL PROTECTED] Subject: Re: [asterisk-users] RE: alcatel ip touch 4068 ... sip? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 2006/7/28, Leo Ann Boon [EMAIL PROTECTED]: (AstATN) wrote: Hi Cesc, Alcatel does not offer any SIP Terminal, it phone's are H.323 it's handle for their own features usages. ( like ADSI type ) Common misconception. Their phones are not H.323 despite claims in their documentation. The server has to do the signaling conversion. The native protocol is UAIP (User Agent IP) which runs over UDP. Hi, I've never heard of that (UAIP) before ! Do you have anything describing this protocol ? Would it be difficult to implement it inside Asterisk just like UNISTIM has been ? Regards -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/77bddf da/attachment-0001.htm -- Message: 13 Date: Fri, 28 Jul 2006 10:55:58 +0200 From: Joseph Dudash [EMAIL PROTECTED] Subject: [asterisk-users] asterisk+ooh323.. one way audio issue To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi guys, I tried to make call from SIP channel to H323 using asterisk+ooh323. The SIP client is x-lite. The problem is that there is one way audio. I hear everything from h323 endpoint, and I see the messages also: Got RTP packet from 66.135.35.xx:5002 (type 3, seq 36250, ts 74400, len 33) Sent RTP packet to 212.183.41.xx:45956 (type 18, seq 22288, ts 70880, len 20) But the problem, when I talk via X-lite, or send dtmf tones, no audio is transfered, no RTP packets on asterisk console. My ooh323.conf: [general] port=1720 bindaddr=0.0.0.0 gateway=no h323id=ObjSysAsterisk e164=100 callerid=asterisk gatekeeper = DISABLE disallow=all allow=g729 allow=gsm allow=ulaw Note that I tried all combinations of faststart and h245tunneling, but no luck. Also tried with gsm and g729 codecs (that time X-pro was used) but same oneway audio. Asterisk version is 1.2.7.1 With full debug this is what I see in asterisk console: Jul 28 10:40:22 DEBUG[15775]: pbx.c:1674 pbx_extension_helper: Launching 'Dial' Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable STACK-test-381637790067-2. Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable STACK-test-381637790067-1. Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPCALLID. Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPURI. Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel OOH323/66.135.33.xx-14b0 to read format gsm Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel SIP/66-9cbb to write format gsm Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel SIP/66-9cbb to read format gsm Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel OOH323/66.135.33.xx-14b0 to write format gsm Jul 28 10:40:22 DEBUG[16193]: res_config_mysql.c:636 mysql_reconnect: MySQL RealTime: Everything is fine. Jul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set
Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
Filip Drągowski wrote: on one cosnole a do asterisk -r on other i do asterisk -rx logger rotate and the result is -- Remote UNIX connection Asterisk Event Logger restarted Asterisk Queue Logger restarted -- Remote UNIX connection disconnected how often new log files are created ? = how many log files are created in 1 second ? Very often, it slows down as the number of log files in the /var/log/asterisk directory increases. there is some kind of regularity or it is done randomly (10logs/1s and another time 20logs/1s) ? Seems to happen more often than not (the past three days) overnight. We're assuming that it starts when asterisk does it's daily log rotate itself and then gets itself into a spin... -- Kenny Millington Systems Developer 3aIT Limited T: 0870 881 5097 F: 01403 248 105 E: [EMAIL PROTECTED] W: http://www.3ait.co.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] registration process
Hi all, I wonder if there are 2 UAs having the same sip account and password. If they both register to the same server in same time. Both of them can register successfully and make calls. Am I right? How can I prevent the above case, say only one UA can register to the server? Please advice. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
asterisk does daily log rotate all along ? i didn't know that it is
posiible
i create file in /etc/logrotate.d/asterisk (copy of postgrsql and
renamed it)
/var/log/asterisk/full {
daily
rotate 10
copytruncate
delaycompress
compress
notifempty
create 640 root root
}
i have 10 last full log, 8 oldest gziped
on one cosnole a do asterisk -r
on other i do asterisk -rx "logger rotate"
and the result is
-- Remote UNIX connection
Asterisk Event Logger restarted
Asterisk Queue Logger restarted
-- Remote UNIX connection disconnected
how often new log files are created ? = how many log files are created
in 1 second ?
Very often, it slows down as the number of log files in the
/var/log/asterisk directory increases.
there is some kind of regularity or it is done randomly (10logs/1s and
another time 20logs/1s) ?
Seems to happen more often than not (the past three days) overnight.
We're assuming that it starts when asterisk does it's daily log rotate
itself and then gets itself into a spin...
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[asterisk-users] Voicmail Question
Hi list, is it possible to pick up a call from VoiceMail system? Didn't find nothing on voip-info.org Thanks for your answers KAI ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MWI from Octel to Asterisk
When we were first looking at Asterisk, I explored some options of integrating our existing Octel voicemail systems with it. The only possible way I could come up with (understanding that I am by no means an Octel expert) was DTMF inband integration. The most difficult part seemed to be fabricating the SIP messages for MWI, but that can be done with sipsak without too much pain. In the end we didn't do that and just used Asterisk's voicemail, so I don't have any working configs for you. But it seemed quite plausible at the time with a bit of effort. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Diehl Sent: Thursday, July 27, 2006 11:01 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MWI from Octel to Asterisk I'm afraid I wasn't very clear on this point. Our customers CURRENTLY have ISDN phones that they are used to and quite happy with. When we roll out VoIP, we will probably replace these phones with VoIP phones. But since our customers are used to the MWI, we need to be sure we can retain that functionality. BTW, the Octel is connected to the 5ESS via T1, as will the Asterisk server. Hope this helps you help me. Thank you, Mike. On Wednesday 26 July 2006 10:57, Olivier wrote: 2006/7/26, Mike Diehl [EMAIL PROTECTED]: We have ISDN phones that have a Message Light that we don't want to break. Hi Mike, How will these phones be connected ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stream file outputs only silence, even with asterisk example gsm file
Hi all! I am trying to hook up a text to speech system to Asterisk via AGI. The AGI script generates a sound file and tells Asterisk to play that file via STREAM FILE. I am creating the sound file in alaw format. My problem is that I do not get any sound output on my softphone (IAX soft phone) AGI Rx EXEC AGI tts.agi|Hello! Welcome to Asterisk!! AGI Tx agi_request: tts.agi AGI Tx agi_channel: IAX2/1234-2 AGI Tx agi_language: en AGI Tx agi_type: IAX2 AGI Tx agi_uniqueid: 1154082448.87 AGI Tx agi_callerid: 10101 AGI Tx agi_calleridname: Guido Sohne AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: 100 AGI Tx agi_rdnis: unknown AGI Tx agi_context: corenett AGI Tx agi_extension: 100 AGI Tx agi_priority: 1 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx STREAM FILE /tmp/tmp4636.92179778618 # AGI Tx 200 result=0 Checking /tmp, we see that the file does exist, which ties in with the 200 response code after the script issues a STREAM FILE command to Asterisk AGI. ls -l /tmp -rw-r--r-- 1 asterisk root 75388 2006-07-28 10:27 tmp4636.92179778618.alaw I've tried to copy one of the original Asterisk sound files into /tmp and then do a STREAM FILE on it. That also does not result in any sound being played! AGI Rx STREAM FILE /tmp/why-no-answer-mystery # AGI Tx 200 result=0 ls -l /tmp -rw-r--r-- 1 guidoguido 10626 2006-07-28 10:31 why-no-answer-mystery.gsm So what gives here? I can't understand what is going wrong and would appreciate some help. -- G. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cmd DIAL - Who picked up the call?
Hi, if I do Dial(SIP/peer1/numberZap/g1/Number) can I somehow figure out who exactly picked up the call? In the cdrs dstchannel I can see the channel but not the extension dialed. E.G. a Dial(Zap/10/43) will result in a CDR Zap/10-1 which does not help me unfortunatly. Any ideas? Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird E1 problem
Hi all, I have a weird problem. I have a Digium te400p with 4 E1s coming in to it. When one of the E1 lines is plugged into any of the four connections on the digium card I get YELLOW / RED alerts when I cat /proc/zaptel/SOCKET. But I can move the bad E1 line into any socket on the digium card and I'll see the error there. I can actually take/make calls on the bad line but there is lots of clicking sounds. Now the weird thing... The Telco has checked the bad line lots of times and left a data analyzer on the premises and the line always checks out 100% ok for them (so they can't/wont fix it). Its really strange how the other 3 lines work perfectly. Does anyone have any ideas? thanks, Derek -- Derek Conniffe Rivertower Ltd DID Number: 01 440 1806 (International: 00 353 1 440 1806) Ireland: (Local) 01 440 1800 United Kingdom: 0870 068 2368 International: 00 353 1 440 1800 Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 440 1801 (International: 00 353 1 440 1801) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Fritz!Box Fon ATA
-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed Hi, I'm starting to use the new PAP2T instead of the old PAP2-NA for my new installations. I'm having a weird problem: when a call is comming from a zaptel channel (from a bri with bristuff driver) the PAP2T say BUSY to the SIP channel. I have disabled all the features like DND and call forward. If it's the last line for this number in the dialplan I can answer the call normally, but I can't use voicemail, because it jump to it each time. I have installed about 50 PAP-NA and never had this kind of problem. If the call is coming from an other PAP2T (via asterisk with canreinvite=no), everything is fine. This occur with asterisk 1.0.10 and 1.2.9.1 the firmware version for the PAP2T is 3.1.9(LSc) I am using a dialplan coming from another customer with a similar setup, but with PAP2-NA, where it's working fine. What can I do to fix this. Regards, Olivier -- Message: 11 Date: Fri, 28 Jul 2006 10:30:50 +0200 From: Olivier [EMAIL PROTECTED] Subject: Re: [asterisk-users] Sip phone settings set when user registers To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 2006/7/27, Nik Engel [EMAIL PROTECTED]: User logs into any phone and the settings of the phone are always the same. Meaning individual key assignement is always the same. Hi, Do you mean : 1. Without user logins, phones are unusable ? Or do you plan to offer default services (local calls for instance) for unidentifed users ? I'm not sure many phones offer special keys for login-logout. 2. What should happen when users change phones settings ? Shall these changes be saved somehow (during logoffs ?) and somewhere for latter reuse ? That implies phone config should be portable from one phone to another. That doesn't seem easy if phones are installed in different locations. Regards -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/b8a1c9 87/attachment-0001.htm -- Message: 12 Date: Fri, 28 Jul 2006 10:52:20 +0200 From: Olivier [EMAIL PROTECTED] Subject: Re: [asterisk-users] RE: alcatel ip touch 4068 ... sip? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 2006/7/28, Leo Ann Boon [EMAIL PROTECTED]: (AstATN) wrote: Hi Cesc, Alcatel does not offer any SIP Terminal, it phone's are H.323 it's handle for their own features usages. ( like ADSI type ) Common misconception. Their phones are not H.323 despite claims in their documentation. The server has to do the signaling conversion. The native protocol is UAIP (User Agent IP) which runs over UDP. Hi, I've never heard of that (UAIP) before ! Do you have anything describing this protocol ? Would it be difficult to implement it inside Asterisk just like UNISTIM has been ? Regards -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/77bddf da/attachment-0001.htm -- Message: 13 Date: Fri, 28 Jul 2006 10:55:58 +0200 From: Joseph Dudash [EMAIL PROTECTED] Subject: [asterisk-users] asterisk+ooh323.. one way audio issue To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi guys, I tried to make call from SIP channel to H323 using asterisk+ooh323. The SIP client is x-lite. The problem is that there is one way audio. I hear everything from h323 endpoint, and I see the messages also: Got RTP packet from 66.135.35.xx:5002 (type 3, seq 36250, ts 74400, len 33) Sent RTP packet to 212.183.41.xx:45956 (type 18, seq 22288, ts 70880, len 20) But the problem, when I talk via X-lite, or send dtmf tones, no audio is transfered, no RTP packets on asterisk console. My ooh323.conf: [general] port=1720 bindaddr=0.0.0.0 gateway=no h323id=ObjSysAsterisk e164=100 callerid=asterisk gatekeeper = DISABLE disallow=all allow=g729 allow=gsm allow=ulaw Note that I tried all combinations of faststart and h245tunneling, but no luck. Also tried with gsm and g729 codecs (that time X-pro was used) but same oneway audio. Asterisk version is 1.2.7.1 With full debug this is what I see in asterisk console: Jul 28 10:40:22 DEBUG[15775]: pbx.c:1674 pbx_extension_helper: Launching 'Dial' Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable STACK-test-381637790067-2. Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable STACK-test-381637790067-1. Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPCALLID
Re: [asterisk-users] IAX2 Connection fails over time...
Stuart Sheldon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey all, I have a x86 Pentium D asterisk system with two Digium 400's in it. I am establishing a IAX2 Connection to another Asterisk system running on a Solaris server. When a call is placed between the two systems, everything seems fine for a variable period of time, then for some reason beyond what my diagnostics has found, the call begins to lag, and both asterisk's servers report LAG. Network wise, the systems have a 4-10 msec ping time. Once the call is terminated, everything returns to normal, and the call can be reconnected. Until the call is terminated, no other calls can be setup with that host. Both systems are running 1.2.x. We are using the GSM codec for the calls. Any ideas on what we should check? Try changing the codec for the iax link to g726 and report back. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel trunk failed to compile
Morning everybody, I try to install an asterisk test server with trunk branch and get this error when compiling zaptel. Asterisk core compile fine as well as SVN 1.2 branch. It's a Debian SARGE running on 2.4.27 kernel. zttranscode.c: In function `zt_tc_mmap': zttranscode.c:378: warning: passing arg 1 of `remap_page_range_R69d01e73' makes integer from pointer without a cast zttranscode.c:378: error: incompatible type for argument 4 of `remap_page_range_R69d01e73' zttranscode.c:378: error: too many arguments to function `remap_page_range_R69d01e73' make[1]: *** [zttranscode.o] Error 1 make[1]: Leaving directory `/usr/src/zaptel-trunk' Thanks for any advise. -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
I use logrotate too, because I didn't know of the functionality in Asterisk.
Logrotate works fine for me though.
Kenny, you should give it a try!
K
On 7/28/06, Filip Drągowski [EMAIL PROTECTED] wrote:
asterisk does daily log rotate all along ? i didn't know that it is posiiblei create file in /etc/logrotate.d/asterisk (copy of postgrsql and renamed it)/var/log/asterisk/full {
daily rotate 10 copytruncate delaycompress compress notifempty create 640 root root}i have 10 last full log, 8 oldest gziped
on one cosnole a do asterisk -r
on other i do asterisk -rx logger rotate
and the result is
-- Remote UNIX connection
Asterisk Event Logger restarted
Asterisk Queue Logger restarted
-- Remote UNIX connection disconnected
how often new log files are created ? = how many log files are created
in 1 second ?
Very often, it slows down as the number of log files in the
/var/log/asterisk directory increases.
there is some kind of regularity or it is done randomly (10logs/1s and
another time 20logs/1s) ?
Seems to happen more often than not (the past three days) overnight.
We're assuming that it starts when asterisk does it's daily log rotate
itself and then gets itself into a spin...
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[asterisk-users] Re: bugs.digium.com
This is not a bug. It is just the way it works. The sip debug output is verbose output in asterisk console terminology. Also, the verbose setting in logger.conf has no effect for the console in logger.conf. Printing verbose output is only controlled by the set verbose CLI command. I do not think that this is true. If I turn on sip debug, it doesn't matter what I set set verbose to, it will still go to console. I tried it with set verbose 1 and set verbose 0. either way, it still went to console. -- -- Steven http://www.glimasoutheast.org Russell Bryant [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Thu, 2006-07-27 at 08:32 -0600, Douglas Garstang wrote: I opened bug #0007490 the other day. The issue was that when you do a 'sip debug' on the Asterisk console, there was no way to have this output go _only_ to the messages file. Someone with the id of 'russell' in his infinite wisdom has deemed that this isn't a bug, closed it, and given me -2 karma points. The mantis user russell would be me. Let me introduce myself. I have been an active Asterisk developer for about two years. When I first got involved in Asterisk development, almost nobody ran any release of Asterisk. Everyone ran the code straight out of the development tree in CVS because that was the best that was available. In the Fall of 2004, Mark Spencer asked me to take on the responsibility of managing bug fix releases of Asterisk after he released Asterisk 1.0. At that point, I monitored every change that Mark committed into the development branch of Asterisk, and manually merged bug fixes into the 1.0 branch. I did this for about a year, until the 1.2 release was made. During this time, I created all of the 1.0.X releases. Since Asterisk 1.2 has been released, the development team has taken more of a group responsibility of committing the bug fixes into the release branch, which has resulted in higher quality releases, with much less of a possibility of anything getting missed. However, I am still considered the Asterisk release maintainer and make decisions about what gets included in the release branch when such decisions need to be made. I have fixed countless bugs, added new features, and reviewed and committed hundreds of contributions from the community of developers. Needless to say, I have some experience in the process of Asterisk development. It clearly is a bug, or at the VERY least, a limitation that needs to be fixed. So why the hell did he give me -2 karma points and say 'not actually a bug'. It is not a bug. This is exactly how it is intended to work in the current code. I'm sorry if it was confusing to you. However, the bug tracker is not the appropriate place to go when you are confused about configuration. Fine... so how do you file an enhancement request then? If there's no way to file an enhancement request, then this is the most appropriate place to file this. The bug tracker is really is not a good place for feature requests. You have to understand that this is the tool we have for managing all of Asterisk development contributions and bugs. If every user posted every feature they think should be implemented on there, it would make it much more difficult for us to manage. The developers *do* monitor the mailing lists. One of the major reasons we monitor the lists is to understand the issues that users are facing. Believe me, if you start a discussion on this mailing list regarding this issue, it will get noted, and if we are able, we will make improvements to make things easier for you. Also keep in mind that there are many thousands of users with many different ideas and drastically fewer developers that can implement them. Its damn irritating not being able to have 'sip debug' output go to a file only, and this is what the options in logger.conf imply you should be able to do, which is another reason I don't understand why he took this irrational action. I'll look over the text in logger.conf to see if I can make some things more clear. I will also be thinking about potential ways to implement this new feature. But, keep in mind that this is just another feature request in quite large pool of them. In the future, if you are unsure of how to do something, if something is even possible, or are confused about configuration options, it is much more appropriate to start a discussion here. Then, the details of what is going on, and what could be implemented to make things better can be worked out. We will see this discussion and make note of it. I am very passionate about my work on Asterisk. I fix bugs, implement new features, and do my best to improve the experience of every Asterisk user. Everything I do is in my mind what I have to do to ensure that I am as productive as I can be to improve Asterisk. In a PRODUCTION environment, you can't be running a sip debug to your console. I'm not
[asterisk-users] Re: gxp-2000 configure line appearances
Thanks this was exactly what I was looking for. Thanks Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grand stream 2000 will not dial *xx
I just bought a grand stream 2000. It appears that it will not dial any number with a leading * (*70,*71) So I can not dial any of my Apps in * Can anyone point me in the right direction? Thanks, Rich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Flash operator panel
Can anybody steer me in the right direction? I have installed the fop and have it working okay, first problem is agent logins not changing the state color when an agent logs in. I configured it on two boxes one works the other doesn't, same configs alll the way. The other is more of me not understanding how it works. I only see the buttons that i have programmed and am unable to get the password entry box and can't figure out how to do transfers. Jordan Novak Communications Technician ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Grand stream 2000 will not dial *xx
I just bought a grand stream 2000. It appears that it will not dial any number with a leading * (*70,*71) So I can not dial any of my Apps in * What firmware version are you running? We have plenty of GXP2000s in clients' premises with plenty of numbers beginning * without any problems. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
I am trying to have thier PC run thru the port on the phone and the phone give prioroty to itself and the rest to the PC. When my client does a big download the phone call gets real bad. The docs from SNOM on TOS (or DIFFSERV) is poor and I dont understand it well enough. Anyone have configs or docs on how they did this ? Doid - Original Message - From: Koopmann, Jan-Peter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 28, 2006 3:17 AM Subject: RE: [asterisk-users] SNOM 360 On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
Also SNOM says by Vlan to set the vlan and then the value for the qos. When you set Vlan to 0 it is supposed to be no Vlan. However once I set it the vlan on the SNOM to 0 and I reboot the phone is no long accessable from the network and I have to reset it. Dovid - Original Message - From: Koopmann, Jan-Peter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 28, 2006 3:17 AM Subject: RE: [asterisk-users] SNOM 360 On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cmd DIAL - Who picked up the call?
What about DIAL ( |M(macro-name)) and set the userfield in cdr during execution, ... http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Hi, if I do Dial(SIP/peer1/numberZap/g1/Number) can I somehow figure out who exactly picked up the call? In the cdrs dstchannel I can see the channel but not the extension dialed. E.G. a Dial(Zap/10/43) will result in a CDR Zap/10-1 which does not help me unfortunatly. Any ideas? Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 Connection fails over time...
On Friday 28 July 2006 07:51, Rich Adamson wrote: Any ideas on what we should check? Try changing the codec for the iax link to g726 and report back. Offhand, what are you suspecting? GSM's a pretty light codec in terms of CPU, and he's not lacking in the CPU department either (if this is a small number of concurrent calls and the box isn't doing anything else). -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendtext or sip message - where in RFC
I was looking in apps/sendtext.c hoping to find a reference to the RFC number and section etc where this is talked about. Can someone point me where that information is for a SIP message? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Install asterisk-bristuff for Debian Linux
Following a discussion on this list about a week ago I downloaded and installed Debian Linux. Now I want to install asterisk-bristuff. How do I do that? Better yet, what do I put in /etc/apt/sources.list so I can do apt-get install asterisk-bristuff -- Thanks for your help, Cosmin Prund ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX Inbound Route
You could setup a ring group that included all extensions in your inbound route, the default for freepbx is to have an anydid/anycid route so any calls coming in will be sent to whereever you say (see the inbound routes link in freepbx). You will need to install the ring groups module from the modules section (Tools, Modules) to have this capability.On 7/28/06, Giedrius Augys [EMAIL PROTECTED] wrote: Hi, I have SIP trunk. And I also have a lot of SIP clients. If I want to call from SIP trunk to the Asterisk SIP client, I need to create Inbound route for each endpoint. Maybe is possible to create an endpoint group, because I have a lot of SIP endpoints, and it takes a lot of time to create inbound routes. Or maybe it's only one way to do that. Thanks ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
Koen Van Impe wrote:
I use logrotate too, because I didn't know of the functionality in Asterisk.
Logrotate works fine for me though.
Ok, I believe I see the problem here!
I was told (apparently erroneously) that asterisk does rotation itself
because they didn't rotate before and now they do.
I've just looked in the /etc/logrotate.d/ directory and there's an
asterisk file containing:-
# cat /etc/logrotate.d/asterisk
# system-specific logs may be configured here
/var/log/asterisk/* {
daily
postrotate
/usr/sbin/asterisk -rx logger rotate
endscript
}
Now... If I were to guess I'd guess that the * is matching the logs that
have already been rotated and rotating them, generating yet more files
to be matched by the * and hence rotated... Does that sound plausible?
At any rate, I'm going to specify the files without using a wildcard
match and see how that goes.
--
Kenny Millington
Systems Developer
3aIT Limited
T: 0870 881 5097
F: 01403 248 105
E: [EMAIL PROTECTED]
W: http://www.3ait.co.uk
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Re: [asterisk-users] Grand stream 2000 will not dial *xx
Turn off the call features in the phone, by default the *70 codes are enable in the phone so that the phone can do call waiting and such. If you want asterisk to do this you need to disable the feature codes in the phone. On 7/28/06, Chris Bagnall [EMAIL PROTECTED] wrote: I just bought a grand stream 2000.It appears that it will not dial any number with a leading *(*70,*71) So I can not dial any of my Apps in *What firmware version are you running? We have plenty of GXP2000s in clients' premises with plenty of numbers beginning * without any problems.Regards,Chris--C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change the from@ using the voicemail.conf
Hi, I'm trying to setup the voicemail.conf to email messages, but my mail server fails because the from user is [EMAIL PROTECTED] Does anybody know away to change the user part from root? I'm using exim4 to send the emails. Thanks, Dean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk VOIP / Mikrotik
have a 10 mb ethernet connection from my ISP into ether1 on a PC - Mikrotik 2.9.23 installed. ether2 is the rest of my network behind the router. How do I prioritize packets such that VOIP calls ALWAYS get a "clean channel" through to my Asterisk server, which resides behind that router ? Things sound choppy at best at the moment. HelP! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install asterisk-bristuff for Debian Linux
Google is Your friend http://peen.net/2006/04/15/asterisk-1271-and-zaptel-125-for-debian-sarge/ Following a discussion on this list about a week ago I downloaded and installed Debian Linux. Now I want to install asterisk-bristuff. How do I do that? Better yet, what do I put in /etc/apt/sources.list so I can do apt-get install asterisk-bristuff -- Thanks for your help, Cosmin Prund ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
It has been several years since I had to address similar situations, but I used TUT Systems devices back then. worked great. There are several DSL variants which should work ok. On Jul 27, 2006, at 6:02 PM, Manrique Feoli wrote: another thought, if you are in a bowl, all you need to find is line of sight to one common place from both ends, and place a repeater there. (you could also set two or three steps repeating the signal within points which have line of sight). I'm not sure but I think one repeater would be much cheaper than 20.000ft of copper + extenders + poles+ maintenance, lighning... (even thought you are in Copper Mountain !!, BTW nice spot ). if in the end you decide to go with ethernet, just beware of lighning!!! Brian Vincent (C) escribió: I know.. I know… fiber would be ideal. We have single-mode all over the place. We even have some dark, unterminated strands within 2000ft of this location – it makes me want to cry. Unfortunately lighting it up isn’t an option – we wouldn’t gain anything because we couldn’t connect to anything else to get us the last stretch. Trenching 2000ft isn’t an option – this is National Forest land and we’re not allowed to do that. As far as wireless – no line of sight. This location sits in a little bowl at 11,200’. So what I’m left with is a 400pr, 22awg out to 3000’. Then we jump on 200pr, 24awg aerial cable strung on the 3^rd longest high-speed quad chairlift (10,800’ run). The last leg involves a short underground to another high-speed quad and down 6000’. We can stick a powered repeater in the motor room of the first lift (so I guess a bit further than the original 12,000’ I was thinking.) Yes, we do strange things. If you’re really curious, here’s a map of the campus environment we maintain: http://www.skireport.com/colorado/copper/trailmap/ --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- *From:* [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] *On Behalf Of *Bruce Reeves *Sent:* Thursday, July 27, 2006 4:03 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] long distance ethernet Asterisk I would really look towards fiber, the bandwidth and distance can easily be handled. On 7/27/06, *Manrique Feoli* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, *Brian Vincent (C)* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Two questions: 1. We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. 2. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _ _ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. _ _ ___ --Bandwidth and Colocation provided by Easynews.com http:// easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - --- ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Change the from@ using the voicemail.conf
Hi Dean, In the voicemail.conf, in the [general] section near the top, I've got ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] My e-mails now come from [EMAIL PROTECTED], making to easy to set up a filter in my e-mail client to move voicemail messages into a specific folder HTH Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: 28 July 2006 14:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change the from@ using the voicemail.conf Hi, I'm trying to setup the voicemail.conf to email messages, but my mail server fails because the from user is [EMAIL PROTECTED] Does anybody know away to change the user part from root? I'm using exim4 to send the emails. Thanks, Dean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/402 - Release Date: 27/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which card do you recommend for heavy load application?
I'm thinking to implement an application that may need 120 channels (4 E1 spans) being recorded in WAV49 format simultaneously, with echo cancellation, etc. What card would you recommend for this kind of load? (independently of the underlying hardware, assume the best possible). I've tested a Digium TE407P, and it behaves ok for about 80 channnels in an ordinary PC, but I haven't been able to test a Sangoma card in the same condition. So, in your expert opinion, which will be the best choice for this case? Digium TE407P Sangoma A104d AFT Thanks for your recommendations. -- Atly. Alvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendtext or sip message - where in RFC
I was looking in apps/sendtext.c hoping to find a reference to the RFC number and section etc where this is talked about. Because sendtext.c is not SIP specific you will not find a reference to SIP related information there. chan_sip.c has a reference to RFC 3428 (http://www.rfc-editor.org/rfc/rfc3428.txt). Have a look at the comment of the function receive_message() in chan_sip.c. Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] long distance ethernet Asterisk
Brian, While I can't say we've used this specific product, I can say that anything we have used from RAD has been outstanding and highly reliable. http://www.rad-direct.com/App-Ethernet-extender-copper.htm?menuId2=Applicati onMenumenuId=Extenders2 For a season pass or two I'll come help you light it up ;-) Jeremy Porier Senior Director of Information Systems and Technology Colorado Christian University [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Manrique Feoli Sent: Thursday, July 27, 2006 5:02 PM To: Brian Vincent (C); Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] long distance ethernet Asterisk another thought, if you are in a bowl, all you need to find is line of sight to one common place from both ends, and place a repeater there. (you could also set two or three steps repeating the signal within points which have line of sight). I'm not sure but I think one repeater would be much cheaper than 20.000ft of copper + extenders + poles+ maintenance, lighning... (even thought you are in Copper Mountain !!, BTW nice spot ). if in the end you decide to go with ethernet, just beware of lighning!!! Brian Vincent (C) escribió: I know.. I know fiber would be ideal. We have single-mode all over the place. We even have some dark, unterminated strands within 2000ft of this location it makes me want to cry. Unfortunately lighting it up isnt an option we wouldnt gain anything because we couldnt connect to anything else to get us the last stretch. Trenching 2000ft isnt an option this is National Forest land and were not allowed to do that. As far as wireless no line of sight. This location sits in a little bowl at 11,200. So what Im left with is a 400pr, 22awg out to 3000. Then we jump on 200pr, 24awg aerial cable strung on the 3^rd longest high-speed quad chairlift (10,800 run). The last leg involves a short underground to another high-speed quad and down 6000. We can stick a powered repeater in the motor room of the first lift (so I guess a bit further than the original 12,000 I was thinking.) Yes, we do strange things. If youre really curious, heres a map of the campus environment we maintain: http://www.skireport.com/colorado/copper/trailmap/ --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bruce Reeves *Sent:* Thursday, July 27, 2006 4:03 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] long distance ethernet Asterisk I would really look towards fiber, the bandwidth and distance can easily be handled. On 7/27/06, *Manrique Feoli* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, *Brian Vincent (C)* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Two questions: 1. We need to run Ethernet out to a really long distance 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 10Mbps. 2. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. __ ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing
Re: [asterisk-users] Install asterisk-bristuff for Debian Linux
Thanks! It works! (at first) I installed my deb from the given repository and I think it all went find. Asterisk starts up and I can get to the console. But... where are the drivers? updatedb / locate sees no zaptel drivers, and I've got none of the zapp tools on the system. Is that a separate download/install? If so, what's the name of the package I need to install? Thanks Filip Drągowski wrote: Google is Your friend http://peen.net/2006/04/15/asterisk-1271-and-zaptel-125-for-debian-sarge/ Following a discussion on this list about a week ago I downloaded and installed Debian Linux. Now I want to install asterisk-bristuff. How do I do that? Better yet, what do I put in /etc/apt/sources.list so I can do apt-get install asterisk-bristuff -- Thanks for your help, Cosmin Prund ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One extension to ring on multiple outside lines
I have a need to have a single extension actually ring on 2 phone lines which are not extensions (they are analog phone lines). Does anyone know a suitable extensions.conf config for this? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One extension to ring on multiple outside lines
- Original Message - From: Dave Morrow [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 11:34:37 -0300 Subject: [asterisk-users] One extension to ring on multiple outside lines I have a need to have a single extension actually ring on 2 phone lines which are not extensions (they are analog phone lines). Does anyone know a suitable extensions.conf config for this? Sure! exten = 145,1,Dial(Zap/1Zap/2) That line would dial both Zap/1 and Zap/2 whenever someone called 145. The first one to answer gets the call. Is that what you were looking for? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicmail Question
- Original Message - From: Kai Ober [mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:45:22 -0300 Subject: [asterisk-users] Voicmail Question Hi list, is it possible to pick up a call from VoiceMail system? If you mean to grab a call that is currently in the Voicemail application, then no - nothing is currently implemented to do exactly this. There are some things out there that if you put them together they might be able to do this. Like redirecting an active call to another extension/context. If this isn't what you meant, then please do respond with a better explanation. Didn't find nothing on voip-info.org Thanks for your answers KAI Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Grand stream 2000 will not dial *xx
Here is the software version: Program-- 1.1.0.16Bootloader-- 1.1.0.1 When I pick up the line and dial *70 it just disappears and never dials. If I enable early dial it does dial *70 but then it breaks my outbound routes. Thanks Rich I just bought a grand stream 2000. It appears that it will not dial any number with a leading * (*70,*71) So I can not dial any of my Apps in * What firmware version are you running? We have plenty of GXP2000s in clients' premises with plenty of numbers beginning * without any problems. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] registration process
- Original Message - From: unplug [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:35:34 -0300 Subject: [asterisk-users] registration process Hi all, I wonder if there are 2 UAs having the same sip account and password. If they both register to the same server in same time. Both of them can register successfully and make calls. Am I right? They can register to the same account yes but only the last one that registered will get calls directed at the account. As for making calls they'll always be able to unless you have an entry setup that just uses the registered IP address/port for authentication (probably not). How can I prevent the above case, say only one UA can register to the server? Please advice. The only way you *might* be able to block extra registrations is by limiting the account to a specific IP range for registrations but then if you tried to register elsewhere with a legitimate attempt, it would be blocked too. Thanks. Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canreinvite
- Original Message - From: Giordano Grandis [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:01:08 -0300 Subject: [asterisk-users] Canreinvite How can I check if SIP re-invite is really working ? If you do a sip debug you should see two INVITEs to each side after the call is established with the IP address of the GXP2000 in the SDP. You can also run rtp debug to see if the RTP audio stream is running through Asterisk. I'm trying it with two grandstream gxp2000. Thanks Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] One extension to ring on multiple outside lines
Yes, to some extent it is what I want, but I want it to dial outside lines (ie. 800-555-1212 and 800-666-3434) insteand of a Zap channel. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Friday, July 28, 2006 6:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One extension to ring on multiple outside lines - Original Message - From: Dave Morrow [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 11:34:37 -0300 Subject: [asterisk-users] One extension to ring on multiple outside lines I have a need to have a single extension actually ring on 2 phone lines which are not extensions (they are analog phone lines). Does anyone know a suitable extensions.conf config for this? Sure! exten = 145,1,Dial(Zap/1Zap/2) That line would dial both Zap/1 and Zap/2 whenever someone called 145. The first one to answer gets the call. Is that what you were looking for? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grand stream 2000 will not dial *xx
does it reach the asterisk console? and have you turned off the dial features in the phone?On 7/28/06, Cavanna, Richard [EMAIL PROTECTED] wrote:Here is the software version:Program-- 1.1.0.16Bootloader-- 1.1.0.1When I pick up the line and dial *70 it just disappears and never dials.If I enable early dial it does dial *70 but then it breaks my outbound routes.ThanksRich I just bought a grand stream 2000.It appears that it will not dial any number with a leading *(*70,*71) So I can not dial any of my Apps in *What firmware version are you running? We have plenty of GXP2000s in clients' premises with plenty of numbers beginning * without anyproblems.Regards,Chris--C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Grand stream 2000 will not dial *xx
Tom, Disabling the features worked. Thanks. Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer call in SIP
- Original Message - From: Victor Moreno [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Fri, 28 Jul 2006 06:57:48 -0300 Subject: [asterisk-users] Transfer call in SIP Hello, I am running TrixBox. if already in a call session from ZAPTEL to SIP, the user want to transfer the call to a different extension. The user have to dial *extension ? How do you plan on doing the transfer? If your SIP phone has transfer capability built in you should use that. If you're using the transfer capability in Asterisk then you should check how TrixBox has that setup by default (anyone know?). Any configuration is needed to be done in trixbox? Thanks Victor Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install asterisk-bristuff for Debian Linux
I installed the following packages as well: ii libzap-dev 1.0.1-1 Zapata telephony interface library (developm ii libzap11.0.1-1 Zapata telephony interface library (runtime) ii zaptel 1.2.7-1 zapata telephony utilities ii zaptel-source 1.2.7-1 Zapata telephony interface (source code for (the last one must be configured and compiled as a kernel module offcourse) Also see http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+Debian about it. Good luck! Tijl Van den Broeck On 7/28/06, Cosmin Prund [EMAIL PROTECTED] wrote: Thanks! It works! (at first) I installed my deb from the given repository and I think it all went find. Asterisk starts up and I can get to the console. But... where are the drivers? updatedb / locate sees no zaptel drivers, and I've got none of the zapp tools on the system. Is that a separate download/install? If so, what's the name of the package I need to install? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS functionality of bristuff (0.3.0-PRE-1r) with a Junghanns duo GSM PCI card
Hi all, I was quite excited to unearth the gsm send sms channel destination message command in chan_zap.c, but now I've hit a dead end in my efforts to deal with _received_ messages. When my card receives an SM, it prints a... -- SMS received on span 1. PDU: number ...message to the console, but I'm not sure how to get access to that information through any other means. As I see it, my options include... (1) Hacking chan_zap.c (2) Finding an appropriate verbosity level or debugging option that will log such messages to a text file somewhere. I don't mind dealing with the messages through another application (or through AGI scripts), but Asterisk is currently my only means of communicating with the Junghanns. Any suggestions? Thanks! -Chris -- [EMAIL PROTECTED] (PGP key at http://www.aduni.org/~walker/key.html) signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR IP Authorization
- Original Message -
From: Khaled Chehab
[mailto:[EMAIL PROTECTED]
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' [mailto:[EMAIL PROTECTED]
Cc:
[EMAIL PROTECTED]
Sent: Fri, 28 Jul 2006 06:34:05
-0300
Subject: [asterisk-users] CDR IP Authorization
Dear
This function retrieves the ip address of the caller ,I want to import the
value of (recvip) in the mysql cdr ,how can I do that
exten = s,1,NoOp(${SIPCHANINFO(recvip)})
The only place that you could put this to have it stored in the record would be
the user field. Here's an example for storing it there:
exten = s,1,Set(CDR(userfield)=${SIPCHANINFO(recvip)})
Regards
Joshua Colp
Digium
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[asterisk-users] Extending call parking to display park extension on the handset display
I am moving this thread from -dev to -users. The thread is below. Synopisis, I am relatively new to asterisk and even though I've looked through the docs I can not find a way to accomplish what I am trying to do. I am trying to, upon park, send a message to a SIP phone which will display the parking spot number instead of simply 'say'ing it. --I indeed ran into parkandannounce, It seems to be what I want, unfortunately I know nothing about asterisk and even though I downloaded th book I am super lost. I am digging through docs right now, and it seems that you can extend asterisk in all sorts of ways, so I'm thinking hacking into features.c is probably a poor choice considering that the conf files seem to be able to do all sorts of things. I guess my biggest problem ATM is figuring out if / how I can tell my SIP phones to display a custom string on their display. I apologize if this is too basic for this list, just let me know and I can move the discussion to Users, I just assumed that since source code changes are involved it may belong here. The thing is I am just a perl programmer who can hack a little C. one of the other guys here does the phones but he just had surgery and is out and all doped up. So its up to me to get this ball rolling. Guillermo Roditi- Hide quoted text - On 7/27/06, Alexander Lopez [EMAIL PROTECTED] wrote: - Hide quoted text - You can use the SIPpeer functions to grab the IP address of the calling phone you can do this before calling the PARK application. I would also look at . ParkAndAnnounce. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Guillermo Roditi Sent: Thursday, July 27, 2006 5:23 PM To: asterisk-dev@lists.digium.com Subject: [asterisk-dev] Extending call parking to display park extension onthe handset display Hi, Currently our Asterisk set up will 'say' back the parking location of a parked call. I'd like to change this behavior to instead display text on the handset's display. My first instinct was to hack an execl() call to call sipsak and relay the message to the phone, unfortunately, this approach needs the IP address of the phone which doesn't seem to be accessible from ast_park_call(). My next idea was to use the ast_sendtext function I came accross in channels.c however that was also not fruitful. I'd like to ask anyone here if they know how I can display a custom string in the phone's display. I know it wont always works, but as long as it works with our handsets i'm fine with it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Change the from@ using the voicemail.conf
Bad form replying to myself, I know, but it looks like my outlook stripped the carriage return. Should be ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] With the comment on the line above the serveremail line Cheers Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mat Stace Sent: 28 July 2006 14:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Change the from@ using the voicemail.conf Hi Dean, In the voicemail.conf, in the [general] section near the top, I've got ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] My e-mails now come from [EMAIL PROTECTED], making to easy to set up a filter in my e-mail client to move voicemail messages into a specific folder HTH Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: 28 July 2006 14:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change the from@ using the voicemail.conf Hi, I'm trying to setup the voicemail.conf to email messages, but my mail server fails because the from user is [EMAIL PROTECTED] Does anybody know away to change the user part from root? I'm using exim4 to send the emails. Thanks, Dean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/402 - Release Date: 27/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/402 - Release Date: 27/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T always busy on incoming calls with zaptel
- Original Message - From: Olivier MONNET [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 05:11:35 -0300 Subject: [asterisk-users] PAP2T always busy on incoming calls with zaptel Hi, I'm starting to use the new PAP2T instead of the old PAP2-NA for my new installations. I'm having a weird problem: when a call is comming from a zaptel channel (from a bri with bristuff driver) the PAP2T say BUSY to the SIP channel. What's the exact SIP response the PAP2T gives? Might it be possible to get a sip debug on a pastebin so myself and others can examine the full dialog? I have disabled all the features like DND and call forward. If it's the last line for this number in the dialplan I can answer the call normally, but I can't use voicemail, because it jump to it each time. I have installed about 50 PAP-NA and never had this kind of problem. If the call is coming from an other PAP2T (via asterisk with canreinvite=no), everything is fine. This occur with asterisk 1.0.10 and 1.2.9.1 the firmware version for the PAP2T is 3.1.9(LSc) I am using a dialplan coming from another customer with a similar setup, but with PAP2-NA, where it's working fine. What can I do to fix this. Regards, Olivier Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Outbound SIP Trunks
- Original Message -
From: Aaron Anderson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Fri, 28 Jul 2006 03:52:59 -0300
Subject: [asterisk-users] Multiple Outbound
SIP Trunks
I have 3 sip trunks registered with an outside provider, however
asterisk always seems to work when going out the third trunk. Any way
to round-robin this so that we can make more than one outbound call at a
time?
Currently chan_sip has no capability to group together outbound trunks as you
can in zaptel. This is mostly due to the fact that on SIP you don't know that
you can't send someone a call until you send it to them, or you keep track on
your side based on predetermined rules.
You might need to end up using the GROUP capability to limit each trunk to one
call each and have failover to the next.
[macro-call-trunk]
exten = s,1,GotoIf($[${GROUP_COUNT(${ARG1})}=0]?avail:busy)
exten = s,n(avail),Set(GROUP()=${ARG1})
exten = s,n,Dial(${ARG2}||t)
exten = s,n,Hangup
exten = s,n(busy),Noop()
exten = _1NXXNXX,1,Macro(call-trunk|trunk-1|SIP/[EMAIL PROTECTED])
exten = _1NXXNXX,n,Macro(call-trunk|trunk-2|SIP/[EMAIL PROTECTED])
exten = _1NXXNXX,n,Macro(call-trunk|trunk-3|SIP/[EMAIL PROTECTED])
This is just really quickly done...
Basically the macro checks to see if the group count for the trunk is 0, if it
is then the current call is set to use the group (which means the next time
group_count gets called while this call is up, it'll return 1) and the
destination is dialed. Otherwise it gets returned to the dialplan and it tries
the next trunk.
There may be other solutions out there but you could expand on this one so that
a trunk could support, for example, 2 outbound calls at a time. You would just
see if the group_count is equal to 2 and if so jump to busy, otherwise avail.
Thanks in advance,
Aaron
Joshua Colp
Digium
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Re: [asterisk-users] One extension to ring on multiple outside lines
Dave Morrow wrote: Yes, to some extent it is what I want, but I want it to dial outside lines (ie. 800-555-1212 and 800-666-3434) insteand of a Zap channel. Then exten = 145,1,Dial(Zap/g1/18005551212IAX2/[EMAIL PROTECTED]/18006663434) Where g1 is defined in zapata.conf to go to a PRI or FXO line(s) and 'Provider' is properly defined in iax.conf to talk to your provider. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] One extension to ring on multiple outside lines
- Original Message - From: Dave Morrow [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 11:48:48 -0300 Subject: RE: [asterisk-users] One extension to ring on multiple outside lines Yes, to some extent it is what I want, but I want it to dial outside lines (ie. 800-555-1212 and 800-666-3434) insteand of a Zap channel. That depends on how you call outside numbers regularly, I don't know how your system is setup. Provided you are using a technology that provides call progress (PRIs/VoIP Providers) then you can do like so: Dial(SIP/[EMAIL PROTECTED]SIP/18006663434) Dial(Zap/g1/18005551212Zap/g1/18006663434) If you are using something like analog then it's more difficult because under normal circumstances Asterisk can't do call progress on analog so it immediately considers it answered (provided you are using a zaptel analog card). David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [asterisk-users] Canreinvite
Ok, thanks, also if i do not have rtp debug (i'm using asterisk 1.0.9) Hi -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Joshua Colp Inviato: venerdì 28 luglio 2006 12.54 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Canreinvite - Original Message - From: Giordano Grandis [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:01:08 -0300 Subject: [asterisk-users] Canreinvite How can I check if SIP re-invite is really working ? If you do a sip debug you should see two INVITEs to each side after the call is established with the IP address of the GXP2000 in the SDP. You can also run rtp debug to see if the RTP audio stream is running through Asterisk. I'm trying it with two grandstream gxp2000. Thanks Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CSTA support for asterisk
- Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 03:03:13 -0300 Subject: [asterisk-users] CSTA support for asterisk Hi, Can anybody tell me that is their CSTA support for asterisk Due to the fact that nobody seems to know what it is - I'd say no. Can you shed any light on what it is? sanchal Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stream file outputs only silence, even with asterisk example gsm file
- Original Message - From: Guido Sohne [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Fri, 28 Jul 2006 08:19:21 -0300 Subject: [asterisk-users] stream file outputs only silence,even with asterisk example gsm file Hi all! I am trying to hook up a text to speech system to Asterisk via AGI. The AGI script generates a sound file and tells Asterisk to play that file via STREAM FILE. I am creating the sound file in alaw format. My problem is that I do not get any sound output on my softphone (IAX soft phone) Checking /tmp, we see that the file does exist, which ties in with the 200 response code after the script issues a STREAM FILE command to Asterisk AGI. ls -l /tmp -rw-r--r-- 1 asterisk root 75388 2006-07-28 10:27 tmp4636.92179778618.alaw I've tried to copy one of the original Asterisk sound files into /tmp and then do a STREAM FILE on it. That also does not result in any sound being played! AGI Rx STREAM FILE /tmp/why-no-answer-mystery # AGI Tx 200 result=0 ls -l /tmp -rw-r--r-- 1 guidoguido 10626 2006-07-28 10:31 why-no-answer-mystery.gsm So what gives here? I can't understand what is going wrong and would appreciate some help. Have you tried grabbing an audio file and listening to it outside of Asterisk to confirm it's okay? Have you tried listening to it just in the dialplan and not using an AGI at all for streaming it back? We need to eliminate some variables here and narrow down where the issue might be. -- G. Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One extension to ring on multiple outside lines
Dave Morrow wrote: I have a need to have a single extension actually ring on 2 phone lines which are not extensions (they are analog phone lines). exten = 145,1,Dial(Zap/1Zap/2) That line would dial both Zap/1 and Zap/2 whenever someone called 145. The first one to answer gets the call. Is that what you were looking for? Yes, to some extent it is what I want, but I want it to dial outside lines (ie. 800-555-1212 and 800-666-3434) insteand of a Zap channel. Joshua Colp gave you the general idea of what you needed. Just expand it like you would for any other outgoing call using '' between each line or phone you want to dial. Something like: exten = 145,1,Dial(Zap/g1/18005551212Zap/g1/18006663434) see the wiki for details: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Message waiting question...
Hi On 7/27/06, Joshua Colp [EMAIL PROTECTED] wrote: I don't believe there's anything configurable but if you open app_voicemail.c there's two declarations, VOICEMAIL_DIR_MODE and VOICEMAIL_FILE_MODE which set the permissions. DIR mode is at 0770 right now and FILE mode is at 0660. Hum.. Weird then, on my maching the file mode is definitely 0600 .. I used the ATrpm package for Fedora Core 5... JY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Change the from@ using the voicemail.conf
Thanks, that worked great. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mat Stace Sent: 28 July 2006 14:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Change the from@ using the voicemail.conf Hi Dean, In the voicemail.conf, in the [general] section near the top, I've got ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] My e-mails now come from [EMAIL PROTECTED], making to easy to set up a filter in my e-mail client to move voicemail messages into a specific folder HTH Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: 28 July 2006 14:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change the from@ using the voicemail.conf Hi, I'm trying to setup the voicemail.conf to email messages, but my mail server fails because the from user is [EMAIL PROTECTED] Does anybody know away to change the user part from root? I'm using exim4 to send the emails. Thanks, Dean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/402 - Release Date: 27/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] comcast info -- somewhat offtopic
-Original Message- From: Derek Whitten [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 12, 2006 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] comcast info -- somewhat offtopic Martin Joseph wrote: On Jul 12, 2006, at 7:45 AM, Derek Whitten wrote: A comcast representative told me the other day they are planning on doubling their internet speed from 8Mb to 16Mb at the end of this year. They certainly don't deliver anywhere near 8Mbits per second here... So I don't know what those kind of promises mean. I had about 4 times the bandwidth when it was an @home connection. All down hill since. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users i just upgraded to 8M and my avg d/l speed went up to between 850KB/s - 1.05MB/s I signed up when @home first came out in the Baltimore/Washington area and there were hardly any people sharing bandwidth and no caps in either direction. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wav49 for voicemail attachment not playing
I'm trying to use the wav49 attachment, but it will not play on my machine. I'm running windows xp with media player 10, it comes up with codec 'Microsoft GSM 6.10' not available. Microsoft stated that the GSM 6.10 is included in media player 10. Has anybody else had this problem? Could it be the asterisk not compressing it right, the wav file works fine, but the size is really to big. Thanks, Dean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk cdr shows FAILED
Hi, I'm having trouble in that my asterisk cdr is showing a lot of calls failing. The asterisk cdr shows disposition FAILED, and the last app is: DialIAX2/[EMAIL PROTECTED]/12125551234 I removed my username and changed the phone number there. Any idea what causes this, and how I can troubleshoot it? I also keep getting this notice in the asterisk command line. Does anyone know what it is? Jul 28 08:43:16 NOTICE[10423]: channel.c:2424 __ast_request_and_dial: Don't know what to do with control frame 15 I'm using IAX with Asterisk 1.2.9.1 and am using the call manager api to set up the calls. thanks, Cory -- web: corybantic.us ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Source Directory of ASterisk
Hi, I am using TriBox 1.1.1/Asterisk. I want to know where I can find source directory of Asterisk in system so I can install Asterisk audio conversion module (http://redice.krisk.org/res_conv-0.1.tgz) to convert ulaw prompts into g729 prompts. It requires to point Asterisk source Include directory. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CSTA support for asterisk
http://www.google.com/search?hl=enq=define%3A+CSTAbtnG=Google+Search On 7/28/06, Joshua Colp [EMAIL PROTECTED] wrote: - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 03:03:13 -0300 Subject: [asterisk-users] CSTA support for asterisk Hi, Can anybody tell me that is their CSTA support for asterisk Due to the fact that nobody seems to know what it is - I'd say no. Can you shed any light on what it is? sanchal Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash operator panel
Hi Jordan, You might want to subscribe to FOP mailing list. You can do that from http://www.asternic.org Can anybody steer me in the right direction? I have installed the fop and have it working okay, first problem is agent logins not changing the state color when an agent logs in. I configured it on two boxes one works the other doesn't, same configs alll the way. The other is more of me not understanding how it works. I only see the buttons that i have programmed and am unable to get the password entry box and can't figure out how to do transfers. Agent logins work depending on the type of login that you use. You can use agentlogin, agentcallbacklogin or addqueuemember in asterisk. Each one has a special treatment/config setting in FOP. You can read it in the example config files or the online documentation. About your problems with the security code box, I do not understand what your problem is. You have to enter the security code at least one (and it has to match the one defined in op_server.cfg) in order to perform any action, including transfers. Once that the security code is verifies, the lock icon shows closed and you can perform the action. To transfer a call you have to drag the phone icon to the destination. Anyways, please check FOP archives or subscribe to the mailing list as this is related to FOP and not Asterisk itself. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: bugs.digium.com
-Original Message- From: Steven [mailto:[EMAIL PROTECTED] Sent: Friday, July 28, 2006 6:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: bugs.digium.com This is not a bug. It is just the way it works. The sip debug output is verbose output in asterisk console terminology. Also, the verbose setting in logger.conf has no effect for the console in logger.conf. Printing verbose output is only controlled by the set verbose CLI command. I do not think that this is true. If I turn on sip debug, it doesn't matter what I set set verbose to, it will still go to console. I tried it with set verbose 1 and set verbose 0. either way, it still went to console. Here! Here! :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
I would be surprised if the problem is at the phone. I have nearly a hundred 360s, 190s and not one of them suffers from that problem in the default setting. The phone handles it automatically. BUT..if I download from an external site and I pipe the call over the internet without setting any traffic shaping on the router then it gets jumpy. Also, you may experience the same problem if you're somehow saturating the network interface on the switch or the asterisk server (both which is highly unlikely). Check you have some sort of traffic shaping on your router and ensure you have a decent switch. I like m0n0wall for routers and cisco for switches. -- Message: 9 Date: Fri, 28 Jul 2006 09:08:17 -0400 From: Dovid Bender [EMAIL PROTECTED] Subject: Re: [asterisk-users] SNOM 360 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=Windows-1252; reply-type=original I am trying to have thier PC run thru the port on the phone and the phone give prioroty to itself and the rest to the PC. When my client does a big download the phone call gets real bad. The docs from SNOM on TOS (or DIFFSERV) is poor and I dont understand it well enough. Anyone have configs or docs on how they did this ? Doid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
