[Asterisk-Users] Asterisk Receptionist
Updated asterisk Receptionist. Shouldnt have problems with IAX calls causing an error. http://www.easyhomenetworks.com/AstRec/ Kyle
Re: [Asterisk-Users] Simultaneous ring internal extension and external phone number?
On Wed, Jun 02, 2004 at 02:01:49PM +1000, Shaun Ewing spake thusly: exten = xx,1,Dial(IAX2/[EMAIL PROTECTED]/phoneSIP/phone|60|r) By that example, you can see that I am dialing IAX2/[EMAIL PROTECTED]/phone and SIP/phone at the same time with ring back with a timeout of 60 seconds. Note however that this WILL NOT work if one of the devices you are calling is on a Zap channel. I have a PRI and I would love to ring my cell phone AND my desk phone (SIP) at the same time but if I try only the Zap interface rings. I posted regarding this a few days ago. It seems silly to have to go out through another VOIP provider when I have my own PRI. I have clients who want this feature too so I would really like to solve this problem. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgp33TaAFowNN.pgp Description: PGP signature
RE: [Asterisk-Users] RE: H323
Thanks, Andy. I have thus tried to use the other H323 driver written by Michael, I have used the newest PWLIB and OPENH323 libraries and newest OH323 driver. After installing, I was able to get two way audio and all. I have tried this driver before but at the time, there was a false answer supervision problem and I had to abandon it. Now, it seems that this problem has been resolved. However, now I have another problem, I have always configured to write the cdr on MYSQL. However, now with this driver, I tested inbound sip , outbound sip, no problem with MYSQL, I tested inbound sip, and outbound OH323, cdr has been written onto MYSQL, but when I used inbound OH323 and outbound whatever, then CDRs have NOT been written onto MYSQL. Somehow, after using OH323, cdr is not being written onto MYSQL. Please help, Michael, do you know why please? Thanks TC -Original Message- From: Rechenberg, Andrew [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 5:45 PM To: [EMAIL PROTECTED] Cc: T. Chan Subject: RE: [Asterisk-Users] RE: H323 I am having a similar problem with one-way audio from an Avaya hardphone calling a SIP soft phone. Audio from the hardphone is heard on the receiving end (SIP), but audio is not heard on the hardphone. I know audio is being injected into the sound card and being processed by the SIP client (I've tried both X-Lite and Windows Messenger 4.7.2009) because the audio meters show signal coming into the client however nothing is heard on the other end. I am using the following: CVS-HEAD 5/21/04 Pwlib-1.5.2 Openh323-1.12.2 Regards, Andy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan Sent: Tuesday, June 01, 2004 1:25 PM To: Dmitry Mishchenko; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: H323 Dear All, Thanks, but I was already using a pre May 25 CVS version. Does anyone else have any idea please? Thanks TC -Original Message- From: Dmitry Mishchenko [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 6:22 AM To: [EMAIL PROTECTED]; T. Chan Subject: Re: [Asterisk-Users] RE: H323 On Tuesday 01 June 2004 00:56, T. Chan wrote: Dear All, I have used Asterisk for a few months and I have been using a January CVS version, it has been working but has been regularly crashing. I use Asterisk mostly as a softswitch application receiving H323 calls from customers and send to H323 carriers. Since I have been using an older CVS version, the OpenH323 and Pwlib libraries in use have been 1.11.7 and 1.4.11 respectively. I was thinking of using a current asterisk version and see if it is more stable comparing to the version in use. I upgraded to new version, and I understand that with the new version and the H323 code, I need to use the 1.12.2 and 1.5.2 versions of the OpenH323 and Pwlib libraries respectively. I have, therefore, erased the whole Pwlib and Openh323 folders, recreated with the new versions and did the ./configure.make clean.make opt procedures to compile the drivers. I have then compiled all the zaptel, libpri, asterisk as usual, but when I ran the asterisk, I noticed that most calls (calls mostly were sent from Cisco AS5300 and Cisco AS5350) were getting one way audio, the calling party was not able to hear anything even the call was connected, I am not sure if the called party would hear anything, but obviously something is not working properly. I have not exactly the same but rather similar issue with the latest cvs-head. There are recent changes in call of on_start_logical_channel() After moving it to MyH323_ExternalRTPChannel::OnReceivedAckPDU it stopped being called in my configuration. As a result I don't get any audio after call established. And with older approach when on_start_logical_channel was called at MyH323Connection::OnStartLogicalChannel it was working fine. This change was done on May 25 so you may try to use older code from CVS before this date. Jeremy saying the latest version approach is fine, but its not working for me :(. Dmitry Can any of your experts out there help please, thanks? TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004
Re: [Asterisk-Users] Syntax for 2 ISDN Cards
On Tue, Jun 01, 2004 at 11:34:28PM +0200, Gunnar Schaller wrote: ...cut When I try to make 3 simultaneous connections from SIP to ISDN the first and second one works, but on the third connection this happens: -- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in new stack chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7501. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time -- Executing Dial(SIP/gunnar-26ea, CAPI/7502:7986:bBYEXTENSION) in new stack chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7502. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Do you have 4 b-channels? (2 Lines with 2 channels) According to your email you are in germany, there you need a 2nd NTBA. Well, I could be wrong at all, just my thoughts. -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Difference between native and 3rd party h323 channel driver ?
Dear Michael I tried using the newest version of your H323 driver, but somehow it seems that it is not hanging up the channels and for some reasons, it is NOT writing my cdr to the mysql database, it was writing properly before. As you can see , the call finished at 2:40:12 but refused to hang up properly until timing out 22 seconds later, please help Jun 2 02:40:12 DEBUG[135181]: chan_oh323.c:2014 oh323_write: OH323/R4096: Pushed 10 bytes into smoother... Jun 2 02:40:12 DEBUG[135181]: channel.c:2560 ast_channel_bridge: Didn't get a frame from channel: OH323/R4096 Jun 2 02:40:12 DEBUG[135181]: channel.c:2630 ast_channel_bridge: Bridge stops bridging channels OH323/R4096 and OH323/L24947 Jun 2 02:40:34 ERROR[135181]: chan_oh323.c:1454 oh323_hangup: OH323/L24947: Failed to hangup channel (timeout). -- Hungup 'OH323/L24947' == Spawn extension (inboundh323, 12124445000, 4) exited non-zero on 'OH323/R4096' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Tuesday, June 01, 2004 1:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Difference between native and 3rd party h323 channel driver ? Robert Rozman wrote: Hi, I'm trying to compile h323 channel driver on cvs Asterisk 1.0 but no success (I get a lot of errors - related to pwlib library). I read in docs that there is also 3rd party h323 channel driver (somehow both even share protion of code?). Asterisk-oh323 was the first H.323 channel driver for Asterisk. The included one is a fork of it, which followed a different approach in the internal design and implementation. Currently, both are following totally independent roadmaps. I wonder what are pros and cons of both drivers ? Should I try to compile native driver ? Some features of asterisk-oh323 (OH323 driver): - Jitter buffer (static or dynamic, with configurable limits). - Configurable number of voice frames per RTP packet. - Inbound call rate limiter (experimental, needs more testing). - Configurable limits for inbound, outbound, simultaneous calls at any given time. - RTCP report generation and handling. Normally, you try both of them and keep the one that makes you happy. Thanks in advance, Robert. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
There's a new version out with some bugs fixed major ones fixed: deadlock on call end, iax thread getting locked out, few contact group list bugs, one on exit crash bug fixed I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Adam Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
On Wed, 2 Jun 2004, Adam Hart wrote: I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Can I recommend you label files with version numbering - this must be about the third ? fourth ? firefly-thirdparty you've released. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
true, it's internally versioned though - look at the build number. But yes, I'll start suffixing a buildnumber on the files. i'm hoping this will be the last release before the magic feature called conferencing, unless this sip registration issue is firefly related -Adam gARetH baBB wrote: On Wed, 2 Jun 2004, Adam Hart wrote: I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Can I recommend you label files with version numbering - this must be about the third ? fourth ? firefly-thirdparty you've released. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Syntax for 2 ISDN Cards
When I try to make 3 simultaneous connections from SIP to ISDN the first and second one works, but on the third connection this happens: -- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in new stack chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7501. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time -- Executing Dial(SIP/gunnar-26ea, CAPI/7502:7986:bBYEXTENSION) in new stack chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7502. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Do you have 4 b-channels? (2 Lines with 2 channels) According to your email you are in germany, there you need a 2nd NTBA. Well, I could be wrong at all, just my thoughts. Yes there are 4 b-channels and yes it's germany :o) There are 2 lines with 2 b-channels each. My Asterisk operates at a internal telephone system. As I wrote I can do 2 simultaneous connections, in this case capi info shows that contr1 has no free channels. Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Adam The link doesn't seems to work. Get back the following: Parse error: parse error, unexpected T_STRING in /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121 Reto There's a new version out with some bugs fixed major ones fixed: deadlock on call end, iax thread getting locked out, few contact group list bugs, one on exit crash bug fixed I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Adam Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
fixed Reto Stauss wrote: Adam The link doesn't seems to work. Get back the following: Parse error: parse error, unexpected T_STRING in /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121 Reto There's a new version out with some bugs fixed major ones fixed: deadlock on call end, iax thread getting locked out, few contact group list bugs, one on exit crash bug fixed I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Adam Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Thanks, is working now. Reto Zitat von Adam Hart [EMAIL PROTECTED]: fixed Reto Stauss wrote: Adam The link doesn't seems to work. Get back the following: Parse error: parse error, unexpected T_STRING in /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121 Reto There's a new version out with some bugs fixed major ones fixed: deadlock on call end, iax thread getting locked out, few contact group list bugs, one on exit crash bug fixed I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Adam Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 fallback
Well here is my example. I have a client, who has lots of work associates who call in from all over the world to conference calls. For these calls, many of them use cell phones because of local telco issues. This company then pays the cell bills for these call ins. The bills are astronomical. They want to host the conference calls via Asterisk, and want people who have any access to IP to call in via VoIP. In many of these locations, these folks have dial-up and need a codec with a very low bandwidth usage. Hello G729. But whether or not you use G729 or any other codec doesn't negate the need for a fail-over method for any codec that is limited by a license usage or any other limit. The functionality needs to be there. When I turned 16 and got my drivers license, I bought a car. That was back before the japanese were importing cars in any numbers, and the american built cars were not of the best quality, but hey, it didn't matter to me because I really didn't know any better. I bought a car with an engine, and 4 wheels which got me from place to place. It even had a radio with 4 speakers (self installed), A/C and heat. It also had alot of rattles, cheap plastic that didn't fit perfectly, the doors had to be lifted a bit when closing or they wouldn't latch, etc. Today I have a car, of VERY nice quality, it's still an american mfg car, but it's no better or no worse now that it's european and japanese counterparts. If you came to me and offered me a car and said, but wait, it's cheap and very inexpensive because it doesn't have a CD player (so we don't have to pay fees for the license to that technology) and it doesn't have anti-lock brakes (again no need to pay patent fees on that) and no traction control, or automatic climate control, or cruise control, or independent suspension. BUT it's cheap and it gets you from point A to point B. Would I buy it? No, because my expectations for a vehicle have been set beyond this. Every once in a while someone on this list comes out and says something to the fact of just do this workaround. In many cases, they are correct, and in a beta test environment, I fully understand the reasoning. However, whether it's wise or not, people are coming up to the point where they need to install, expand, or replace their current PBX systems and they can either choose to go pay the nortel's of the PBX world lots of money, or they can take the iPBX plunge. If Asterisk truly wants to play in the iPBX world, then it MUST support the same features that are coming out in the big players iPBX systems. Telephone is HUGE to almost every business. It must work, and it must be able to perform in the manner they want it to perform. My client has said, We are going to use G729 for our remote clients to save bandwidth, and if we ever run out of licences, it need to complete the call with another available codec. At this point, my choice is to either make asterisk work as the spec requires, or install a different iPBX system. I can't go back and say don't use G729, use GSM instead, because some of their clients won't have the BW to use anything but G729. So, to put Asterisk to work for this client, I really need this functionality, and I suppose if it doesn't come out of the community, I can hire a programmer (I'm too dumb to be a programmer) to do it for me, because it will still be cheaper than taking the client down the nortel, etc. road. But this particular instance set aside, I come from a formally trained engineering background with quite a few years in a very stringent engineering field (in other words, if something is poorly engineered people died), and one of the basic tenants is that a well engineered system was able to able to operate and function in any situation you could expect to see under nominal conditions. This G729 codec failover is something that seems to me to be a possible occurrence, and under nominal operating conditions. I recently spent some time chatting with John Todd about another feature that fits in with this, and that is a bandwidth manager. In other words, you set a maximum bandwidth allowed, and then asterisk will limit incoming and outgoing calls that would overrun that limit. This failover system would apply then in a situation where perhaps normally you might set up a G711 connection, but that would overrun the limit, so instead, you fail over to GSM... I guess what I'm trying to say is that the function is needed and in the end will probably be used for MANY things, but right now, the G729 license limit is a strong candidate. Over-engineering is generally significantly better than under-engineering. Just my thoughts on the matter. -Chris On 05:19 PM 6/1/2004, Kevin Walsh wrote: Chris A. Icide [EMAIL PROTECTED] wrote: On 08:53 AM 6/1/2004, Kevin Walsh wrote: Mike Heininger [EMAIL PROTECTED] wrote: It's a pity ... it would be great to fallback to another
[Asterisk-Users] Re: Multi process of *
I think no. Oliver Vermeulen wrote: Hi , Do anybody know how you can run multi proccess of * on a server ? Thanks, O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Syntax for 2 ISDN Cards
On Wed, Jun 02, 2004 at 09:27:14AM +0200, Gunnar Schaller wrote: ...cut chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7502. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Do you have 4 b-channels? (2 Lines with 2 channels) According to your email you are in germany, there you need a 2nd NTBA. Well, I could be wrong at all, just my thoughts. Yes there are 4 b-channels and yes it's germany :o) There are 2 lines with 2 b-channels each. My Asterisk operates at a internal telephone system. As I wrote I can do 2 simultaneous connections, in this case capi info shows that contr1 has no free channels. Next questions: Can you see any messages from 2nd line/card via isdnlog? Can you call your * via the other cards? What msn's do the two established calls use? Do you try to access a 3rd call with msn's from first line? Is it right that you have only two msn's in the capi.conf -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Syntax for 2 ISDN Cards
Hi Gunnar,
here is our capi.conf for two controllers on two different ISDN lines
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=041120
incomingmsn=*
controller=1
softdtmf=0
context=default
devices=2
msn=041682
incomingmsn=*
controller=2
softdtmf=0
context=assistenza
devices=2
believe it or not, but you can see in the chan_capi source code,
the creation of the lines are activated by parsing the line devices=
so it seems that MUST be the last line of every interface parameters.
With this capi.conf and two passive AVM controllers (one PCI, une USB)
with hacked drivers, we do have random problems, when we have many calls,
our server hangs and we must reboot. Actually we are trying to understand if
problems are on chan_capi and this capi.conf or on then AVM hack.
Please let me know if this syntax works for you.
Bye.
Francesco Sibilla
- Original Message -
From: Gunnar Schaller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 11:34 PM
Subject: [Asterisk-Users] Syntax for 2 ISDN Cards
Hi there,
I searched in mailinglist and in web, but no answer to my problem...
Only this post with no answers:
http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html
I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple
controller support). In my Asterisk-box there are 2 Fritzcards
(module for second card compiled with changes on sourcecode found in
the web). capi info shows:
Contr1: 2 B channels total, 2 B channels free.
Contr2: 2 B channels total, 2 B channels free.
Here a snipplet of my capi.conf:
[interfaces]
msn=7501,7502
incomingmsn=*
controller=1,2
devices=2,2
Is that correct? I also tried devices=4.
When I try to make 3 simultaneous connections from SIP to ISDN the
first and second one works, but on the third connection this happens:
-- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7501. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
-- Executing Dial(SIP/gunnar-26ea, CAPI/7502:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7502. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
The interesting part of extensions.conf:
exten = _,1,Dial,CAPI/7501:${EXTEN}:bBYEXTENSION
exten = _,102,Dial,CAPI/7502:${EXTEN}:bBYEXTENSION
Can anyone tell me how to use the B-channels of the second Fritzcard?
Gunnar Schaller
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RE: [Asterisk-Users] problems with TDM400P
Wim, If ya don't need callerid then add the patch at http://www.nodomain.org/asterisk to zaptel and asterisk directories. I did this for UK callerid and the phone now rings on the first ring of the CO. Bit of a bodge but it works. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wim Kerkhoff Sent: 02 June 2004 06:34 To: Asterisk-users Subject: [Asterisk-Users] problems with TDM400P Hi, We have two of these 4 port FXO cards. However, we are having some problems with incoming/outgoing calls. The latest version of Asterisk/zaptel from CVS is being used. Voicemail, internal SIP - SIP calls between Pingtel xpressa hard phones work terrific, echotest is fine, and so on. The zaptel and wcfxs modules load fine, and show all 8 FXO interfaces in dmesg: - Zapata Telephony Interface Registered on major 196 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXO Module 1: Installed -- AUTO FXO Module 2: Installed -- AUTO FXO Module 3: Installed -- AUTO FXO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXO Module 1: Installed -- AUTO FXO Module 2: Installed -- AUTO FXO Module 3: Installed -- AUTO FXO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) - Following problems have been observed, and are preventing us from dumping our existing Nortel Merdian PBX: 1. echo at beginning of call for several seconds, even with various combinations of echocancel and echotraining in zapata.conf 2. even though multiple incoming lines are connected, only the first ZAP channel is picking up. So if one line is in use, nobody else can call in even though there are other lines free. When in debug mode (-gcvvv) nothing is showing up that there's another call coming in. 3. channels don't always hang up properly - HookState shows as offhook for quite some time. 4. Asterisk Zap channels don't see an incoming call until 2 rings after the existing Nortel PBX sees it. Both people calling in and people answering don't like that. I've gone through whatever documentation and mailing list archives, but haven't been able to find working solutions. Have tried various combinations in zaptel.conf and zapta.conf but no luck yet :-( Ideas anyone? Thanks, Wim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
On Wed, Jun 02, 2004 at 08:14:44AM +0100, gARetH baBB wrote: On Wed, 2 Jun 2004, Adam Hart wrote: Can I recommend you label files with version numbering - this must be about the third ? fourth ? firefly-thirdparty you've released. .. but have firefly-thirdparty.exe be a symbolic link to the latest version? My 2p. Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 403 Forbidden between two softphones on same Asterisk
Hi, I have two softphones connected to an Asterisk stable. I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a 403 Forbidden result, while a Diax97a on the same extension will just report Call disconnected by remote. The same is not true when 2000 calls extension 1000. Extension 1000 will ring, and is also able to pick up. Extension 2000 can also call external parties (routed through another Asterisk box), but again, external parties cannot call extension 2000 (they can call extension 1000, however!). I'm confident that I've made a mistake, but I just don't know where. Anyone have any ideas? Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TDM400P: Sharing IRQS?
In article [EMAIL PROTECTED], Leo Ann Boon [EMAIL PROTECTED] wrote: The new TDM400P with FXO doesn't take up any IRQ. I've 2 boards and both are not using any IRQ. Weird - does that mean they can't provide Zaptel timing like the X100P can? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP vs. SIP :-(
Resolved... canreinvite=no (i've put careinvite :-)) igor On Tue, 2004-06-01 at 19:24, Igor Barsanti wrote: I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid=Me host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of build_route: messages from asterisk then the sip client die ... Stopping retransmission on '[EMAIL PROTECTED]' of Request 104: Found build_route: Record-Route hop: sip:[EMAIL PROTECTED];ftag=as61269cb9;lr=on build_route: Contact hop: sip:[EMAIL PROTECTED] .. ...with H.323 client all works perfectly. What's the problem ??? Igor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Ericsson MD110 PBX
I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more proprietary? Regards Julian Pawlowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Multi process of *
If you are just doing VoIP (i.e. no FXO/FXS Cards involved) you should be able to run up multiple virtual copies of Linux * in VMWare or Virtual PC. Though I guess you would need a pretty pokey machine -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of nicolas Sent: 02 June 2004 08:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Multi process of * I think no. Oliver Vermeulen wrote: Hi , Do anybody know how you can run multi proccess of * on a server ? Thanks, O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users PLEASE READ: The information contained in this email is confidential and intended for the named recipient(s) only. If you are not an intended recipient of this email you must not copy, distribute or take any further action in reliance on it and you should delete it and notify the sender immediately. Email is not a secure method of communication and Nomura International plc cannot accept responsibility for the accuracy or completeness of this message or any attachment(s). Please examine this email for virus infection, for which Nomura International plc accepts no responsibility. If verification of this email is sought then please request a hard copy. Unless otherwise stated any views or opinions presented are solely those of the author and do not represent those of Nomura International plc. This email is intended for informational purposes only and is not a solicitation or offer to buy or sell securities or related financial instruments. Nomura International plc is regulated by the Financial Services Authority and is a member of the London Stock Exchange. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bluetooth headsets/phones.
Has anyone managed to use a bluetooth headset or phone with their install of Asterisk? What I had in mind was either have a headset paired with the server and use that to answer/make calls in some way, or forward the calls to my mobile via bluetooth if that's possible. -- -S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Multi process of *
Hi. Johnson-Perkins, Robert wrote: If you are just doing VoIP (i.e. no FXO/FXS Cards involved) you should be able to run up multiple virtual copies of Linux * in VMWare or Virtual PC. Though I guess you would need a pretty pokey machine User Mode Linux is way better for that use, much more efficient. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between native and 3rd party h323 channel driver ?
The hangup of a channel depends on OpenH323. The driver just initiates the call clearing and wait for a response from the library (through a callback function). That response contains the call clearing reason and the call duration. Of course there is a timeout that ensures that the library will answer in a valid time frame. In general, a timeout is a serious error. I'll check if the same happens with older versions of the OpenH323 library and let you know. Regarding the problem of the CDRs not being written, I don't think this is related to the channel driver. They are handled by Asterisk. Michael. T. Chan wrote: Dear Michael I tried using the newest version of your H323 driver, but somehow it seems that it is not hanging up the channels and for some reasons, it is NOT writing my cdr to the mysql database, it was writing properly before. As you can see , the call finished at 2:40:12 but refused to hang up properly until timing out 22 seconds later, please help Jun 2 02:40:12 DEBUG[135181]: chan_oh323.c:2014 oh323_write: OH323/R4096: Pushed 10 bytes into smoother... Jun 2 02:40:12 DEBUG[135181]: channel.c:2560 ast_channel_bridge: Didn't get a frame from channel: OH323/R4096 Jun 2 02:40:12 DEBUG[135181]: channel.c:2630 ast_channel_bridge: Bridge stops bridging channels OH323/R4096 and OH323/L24947 Jun 2 02:40:34 ERROR[135181]: chan_oh323.c:1454 oh323_hangup: OH323/L24947: Failed to hangup channel (timeout). -- Hungup 'OH323/L24947' == Spawn extension (inboundh323, 12124445000, 4) exited non-zero on 'OH323/R4096' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Tuesday, June 01, 2004 1:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Difference between native and 3rd party h323 channel driver ? Robert Rozman wrote: Hi, I'm trying to compile h323 channel driver on cvs Asterisk 1.0 but no success (I get a lot of errors - related to pwlib library). I read in docs that there is also 3rd party h323 channel driver (somehow both even share protion of code?). Asterisk-oh323 was the first H.323 channel driver for Asterisk. The included one is a fork of it, which followed a different approach in the internal design and implementation. Currently, both are following totally independent roadmaps. I wonder what are pros and cons of both drivers ? Should I try to compile native driver ? Some features of asterisk-oh323 (OH323 driver): - Jitter buffer (static or dynamic, with configurable limits). - Configurable number of voice frames per RTP packet. - Inbound call rate limiter (experimental, needs more testing). - Configurable limits for inbound, outbound, simultaneous calls at any given time. - RTCP report generation and handling. Normally, you try both of them and keep the one that makes you happy. Thanks in advance, Robert. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ./M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Ericsson MD110 PBX
I don't have any direct experience with the MD110's and Asterisk, but I would envisage the MD110 digital phones are very much a proprietary protocol, as with Nortel digital phones, you can't mix and match between different vendors. It may be possible to get Ericsson (as well as Nortel and others) digital phones working with Asterisk, I doubt via ADSI, more likely via Dialogic voice boards from Intel. I know after some digging through intel.com for info on their Dialogic voice boards I found some technical info on the signaling used for Nortel digital phones. To successfully get a Ericsson/Nortel/etc digital phone working with Asterisk you'd need to firstly purchase the Dialogic voice board(s), then write the drivers for the Dialogic board for Linux (or maybe they already exist, I haven't checked), then some more drivers/plug-ins to get the Dialogic and the vendor-specific digital phones working with Asterisk. I imagine for the most part, depending on how many phones you have and budget, it really wouldn't be economically feasible - in the long run I think you'd find replacing the phones with SIP handsets and trying to sell off the old digital handsets to recoup some of the upgrade cost would be the way to go. Actually if memory serves, the main purpose of the Intel Dialogic boards is actually interfacing the PC (ie Asterisk or other software) to the digital ports of the proprietary PBX, rather than directly interfacing the PC to the proprietary digital phones. So for instance if you wanted to smoothly transfer calls between the Asterisk SIP extensions and the Ericsson MD110 handsets with all the caller ID details, or perhaps run a fancy IVR or auto-attendant system accessible to the MD110 handsets via Asterisk then they'd be ideal. Otherwise you have to interface via other digital trunk methods or Analog extensions and may not get access to as many features as you can through the digital extension ports. Even if you can use the dialogic boards directly with the proprietary handsets, I can't see the solution really scaling anywhere near as well as the proprietary digital cards that plug into the MD110 PBX itself. Of course it would be nice to see the Ericsson/Nortel phones recycled for use with Asterisk systems, but at this point in time I'm not sure how feasible this is. I do believe Nortel were working on (or perhaps have now released) a small black-box solution that plugs into the existing proprietary Meridian handset and then plugs into Ethernet to essentially turn the phone into a VoIP handset - not sure if it uses SIP protocol. If Ericsson have a black-box solution like this available, then it might be a feasible approach, depending on the cost per box and the existing network infrastructure, as ideally you'd have the black boxes powered over Ethernet so you can install UPS' in the communications cabinets to ensure the phones and network are available during power outages. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Pawlowski Sent: Wednesday, 2 June 2004 7:28 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk with Ericsson MD110 PBX I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more proprietary? Regards Julian Pawlowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth headsets/phones.
Hi, Has anyone managed to use a bluetooth headset or phone with their install of Asterisk? What I had in mind was either have a headset paired with the server and use that to answer/make calls in some way, or forward the calls to my mobile via bluetooth if that's possible. I can use DIAX with the BT headset as audio device. The new version of DIAX (hope to be ready in one week) accept any Ericsson/SonyEricsson BT phone as dialer/callerid device for DIAX. You can use the BT headset paired with the PC and the BT phone to dial and to display info, like on the DIAX display. In a future version you will be able to answer DIAX from the BT headset too. I will post a message here when the new version will be available for download. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Syntax for 2 ISDN Cards
Hello,
Thanks for your capi.conf! It works great! I made the changes,
restarted Asterisk and made 3 calls with success.
Thanks again,
Gunnar Schaller
Hi Gunnar,
here is our capi.conf for two controllers on two different ISDN lines
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=041120
incomingmsn=*
controller=1
softdtmf=0
context=default
devices=2
msn=041682
incomingmsn=*
controller=2
softdtmf=0
context=assistenza
devices=2
believe it or not, but you can see in the chan_capi source code,
the creation of the lines are activated by parsing the line devices=
so it seems that MUST be the last line of every interface parameters.
With this capi.conf and two passive AVM controllers (one PCI, une USB)
with hacked drivers, we do have random problems, when we have many calls,
our server hangs and we must reboot. Actually we are trying to understand if
problems are on chan_capi and this capi.conf or on then AVM hack.
Please let me know if this syntax works for you.
Bye.
Francesco Sibilla
- Original Message -
From: Gunnar Schaller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 11:34 PM
Subject: [Asterisk-Users] Syntax for 2 ISDN Cards
Hi there,
I searched in mailinglist and in web, but no answer to my problem...
Only this post with no answers:
http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html
I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple
controller support). In my Asterisk-box there are 2 Fritzcards
(module for second card compiled with changes on sourcecode found in
the web). capi info shows:
Contr1: 2 B channels total, 2 B channels free.
Contr2: 2 B channels total, 2 B channels free.
Here a snipplet of my capi.conf:
[interfaces]
msn=7501,7502
incomingmsn=*
controller=1,2
devices=2,2
Is that correct? I also tried devices=4.
When I try to make 3 simultaneous connections from SIP to ISDN the
first and second one works, but on the third connection this happens:
-- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7501. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
-- Executing Dial(SIP/gunnar-26ea, CAPI/7502:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7502. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
The interesting part of extensions.conf:
exten = _,1,Dial,CAPI/7501:${EXTEN}:bBYEXTENSION
exten = _,102,Dial,CAPI/7502:${EXTEN}:bBYEXTENSION
Can anyone tell me how to use the B-channels of the second Fritzcard?
Gunnar Schaller
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[Asterisk-Users] Script to import Master.csv in the MySQL database - a short HowTo
Hi, I hope this can help others, so this is it. Use it at your own risk. I have test it on 3 separate systems without any problem. Take care to edit the following files taking into consideration your own settings. If you have all the CDR info in the Master.csv too, then delete all the data from the 'cdr' table in MySQL before running the script bellow in oder to prevent dupplicate records. In my example, I have the following config: CDR database:asteriskcdrdb CDR table:cdr CVS file: /var/log/asterisk/cdr-csv/Master.csv 1. Create a file named 'impcdr2sql' with the following content: #!/bin/bash # make a copy of the original Master.csv file to Master.csv.mod cp -vf /var/log/asterisk/cdr-csv/Master.csv /var/log/asterisk/cdr-csv/Master.csv.mod # format the file to comply with the MySQL data (delete '' chars when need it) # use a VIM script (nofielddelims.vim) for this purpose ex /var/log/asterisk/cdr-csv/Master.csv.mod -c :source nofielddelims.vim -c :exit # run the MySQL commands from the cmd.sql file mysql cmd.sql 2. Enter the command to make the script executable: chmod 755 impcdr2sql 3. Create a file named 'nofielddelims.vim' with the following content: Delete '' chars at the beginning of the line :%s/^// Delete '' chars at the end of the line :%s/$// Delete '' chars near the ',' char :%s/,/,/g :%s/,/,/g Replace '' by '' :%s///g 4. Create a file named 'cmd.sql' with the following content: use asteriskcdrdb; ALTER TABLE `cdr` ADD `tmp1` VARCHAR(30) DEFAULT x NOT NULL; ALTER TABLE `cdr` ADD `tmp2` VARCHAR(30) DEFAULT y NOT NULL; LOAD DATA INFILE '/var/log/asterisk/cdr-csv/Master.csv.mod' replace INTO TABLE cdr FIELDS TERMINATED BY ',' LINES TERMINATED BY '\n' (accountcode,src,dst,dcontext,clid,channel,dstchannel,lastapp,lastdata,calld ate,tmp1,tmp2,duration,billsec,disposition,amaflags,uniq ueid,userfield); ALTER TABLE `cdr` DROP `tmp1`; ALTER TABLE `cdr` DROP `tmp2`; 5. Keep all the files in the same directory. All you need to do is to run the script: ./impcdr2sql as root or as an user with full rights on the asteriskcdrdb database and cdr table E... voila! All your old data from Master.csv is now in the MySQL database in the correct format (I hope). Please feel free to make any improovments you want. I'm not a Linux expert. Best regards to you all, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Syntax for 2 ISDN Cards
On Wed, Jun 02, 2004 at 09:27:14AM +0200, Gunnar Schaller wrote: ...cut chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7502. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Do you have 4 b-channels? (2 Lines with 2 channels) According to your email you are in germany, there you need a 2nd NTBA. Well, I could be wrong at all, just my thoughts. Yes there are 4 b-channels and yes it's germany :o) There are 2 lines with 2 b-channels each. My Asterisk operates at a internal telephone system. As I wrote I can do 2 simultaneous connections, in this case capi info shows that contr1 has no free channels. Next questions: Can you see any messages from 2nd line/card via isdnlog? Can you call your * via the other cards? What msn's do the two established calls use? Do you try to access a 3rd call with msn's from first line? Is it right that you have only two msn's in the capi.conf My problem is solved by Francesco Sibilla in this thread. I changed my capi.conf and now it works. Isdnlog didn't work on my machine, it needs hisax but I don't want to load it because I read anywhere to not load it with Asterisk. Anyway, thanks for your help. Gunnar Schaller ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Galaxy Voice
Hi Kevin et al, I have raised this as a flag with GV. They are on the case. Do you have a Grandstream or similar? It does work on that when configured to use GV directly so it is working at some level. Mark Kevin said: Thanks for your suggestion. I will give it a try. The other issue I have is that the Galaxy service claims it has call waiting. When one call is up on the Galaxy connection, I get a busy when calling the number, the same with an outbound, only one call at a time. Thanks again, Kevin -Original Message- From: Dr. Rich Murphey [mailto:[EMAIL PROTECTED] Sent: Monday, May 31, 2004 2:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Galaxy Voice If it fails to register, check the sip debug output for: REGISTER sip:216.229.127.40 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060 If you see 0.0.0.0 in the 'Via' line, try using nat=yes externip=your external address in your *global* section at the head of sip.conf. I've searched but haven't been able to find where the value is being set to 0.0.0.0. Cheers, Rich -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Sent: Saturday, May 29, 2004 1:24 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Galaxy Voice Yes, I did a search and have what I think is the correct configuration. I did a google search and I didn't see much. I was successful in getting it to work both inbound and outbound with the exception of the notices and warnings. The config I am using is: [galaxyvoice] nat=yes port=5060 fromuser=12345678 fromdomain=216.229.127.40 username=12345678 type=friend secret=12345678 auth=md5 host=216.229.127.40 ;defaultip=216.229.127.40 reinvite=no canreinvite=no dtmfmode=rfc2833 context=inbound-galaxy qualify=yes disallow=all allow=gsm allow=ulaw callerid=12345678 incominglimit=2 defaultexpirey=60 -Original Message- From: brian k. west [mailto:[EMAIL PROTECTED] Sent: Saturday, May 29, 2004 2:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Galaxy Voice Also I think someone posted a galaxy voice config example on the mailing list a few weeks back.. have you searched google yet? bkw - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 11:04 AM Subject: RE: [Asterisk-Users] Galaxy Voice I deeply apologize for the incorrect statement, thanks for taking the time to point out the error...your help is appreciated. -Original Message- From: brian k. west [mailto:[EMAIL PROTECTED] Sent: Saturday, May 29, 2004 1:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Galaxy Voice First off they are not ERRORS they are NOTICE and WARNING. bkw - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 10:26 AM Subject: [Asterisk-Users] Galaxy Voice Has anyone successfully used Galaxy Voice with Asterisk? I am getting the following SIP errors repeated whether it is or isn't behind NAT. May 29 12:17:11 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 29 12:17:25 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again May 29 12:22:52 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 111 (Critical Request) May 29 12:23:06 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again asterisk2*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
Re: [Asterisk-Users] Asterisk with Ericsson MD110 PBX
Working perfektly over E1 link I have MD110 with 13 E1 link and 3 link is on asterisk over digium card Christopher Lee wrote: I don't have any direct experience with the MD110's and Asterisk, but I would envisage the MD110 digital phones are very much a proprietary protocol, as with Nortel digital phones, you can't mix and match between different vendors. It may be possible to get Ericsson (as well as Nortel and others) digital phones working with Asterisk, I doubt via ADSI, more likely via Dialogic voice boards from Intel. I know after some digging through intel.com for info on their Dialogic voice boards I found some technical info on the signaling used for Nortel digital phones. To successfully get a Ericsson/Nortel/etc digital phone working with Asterisk you'd need to firstly purchase the Dialogic voice board(s), then write the drivers for the Dialogic board for Linux (or maybe they already exist, I haven't checked), then some more drivers/plug-ins to get the Dialogic and the vendor-specific digital phones working with Asterisk. I imagine for the most part, depending on how many phones you have and budget, it really wouldn't be economically feasible - in the long run I think you'd find replacing the phones with SIP handsets and trying to sell off the old digital handsets to recoup some of the upgrade cost would be the way to go. Actually if memory serves, the main purpose of the Intel Dialogic boards is actually interfacing the PC (ie Asterisk or other software) to the digital ports of the proprietary PBX, rather than directly interfacing the PC to the proprietary digital phones. So for instance if you wanted to smoothly transfer calls between the Asterisk SIP extensions and the Ericsson MD110 handsets with all the caller ID details, or perhaps run a fancy IVR or auto-attendant system accessible to the MD110 handsets via Asterisk then they'd be ideal. Otherwise you have to interface via other digital trunk methods or Analog extensions and may not get access to as many features as you can through the digital extension ports. Even if you can use the dialogic boards directly with the proprietary handsets, I can't see the solution really scaling anywhere near as well as the proprietary digital cards that plug into the MD110 PBX itself. Of course it would be nice to see the Ericsson/Nortel phones recycled for use with Asterisk systems, but at this point in time I'm not sure how feasible this is. I do believe Nortel were working on (or perhaps have now released) a small black-box solution that plugs into the existing proprietary Meridian handset and then plugs into Ethernet to essentially turn the phone into a VoIP handset - not sure if it uses SIP protocol. If Ericsson have a black-box solution like this available, then it might be a feasible approach, depending on the cost per box and the existing network infrastructure, as ideally you'd have the black boxes powered over Ethernet so you can install UPS' in the communications cabinets to ensure the phones and network are available during power outages. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Pawlowski Sent: Wednesday, 2 June 2004 7:28 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk with Ericsson MD110 PBX I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more proprietary? Regards Julian Pawlowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax Recognizion without Answer? How to Supress this?
Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri * -- Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are recognized as faxes and * tries to forward the call to fax. I do not answer this calls... == Spawn extension (Amt595xxx-In, 595164, 1) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' -- Starting simple switch on 'Zap/62-1' -- Accepting overlap call from '595457' to '034491' on channel 31, span 2 -- Executing SetVar(Zap/62-1, Out=Zap/g1/) in new stack -- Executing Goto(Zap/62-1, OutDial-LCR|BYEXTENSION|1) in new stack -- Goto (OutDial-LCR,034491***,1) -- Executing SetVar(Zap/62-1, LCR=01081) in new stack -- Executing Goto(Zap/62-1, OutDial-Dial|BYEXTENSION|1) in new stack -- Goto (OutDial-Dial,034491,1) -- Executing Dial(Zap/62-1, Zap/g1/0108103|30|TrH) in new stack -- Called g1/010810344918*** -- Redirecting Zap/62-1 to fax extension -- Hungup 'Zap/1-1' == Spawn extension (OutDial-Dial, fax, 0) exited non-zero on 'Zap/62-1' -- Executing Dial(Zap/62-1, Zap/g1/01081fax|30|TrH) in new stack -- Called g1/01081fax -- Channel 2, span 1 got hangup -- Hungup 'Zap/2-1' What have I to change? Could I supress that? Thanks Felix Deierlein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two FXO Cards answering at different times.
Hi all, Anyone know how put my X101P cards to answer at different ring times ? Like x101P(a) Answer at 3 rings x101p(b) Answer at 4 rings My * it's connected into a PBX thats when receive a call send to two lines at same times a ring. (So i must have a way to just put one channel to answer not both at same time) The behavior is that when i answer one the other channel answer too. Then my client receive 2 calls , one its the normal call the other just a signal of busy. Did anyone know how stop this problem ?? (Don't say buy a new PBX ok!!) ;) Thanks alot . Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Controlling SIP mobile extensions.
Hi everybody, I'm trying to develop a web application for controlling if SIP users are registered in * or not, and show it in a web. My problem is that I don't now if it's possible to do a Shell Script to control this: 1. Connect to console. 2. Execute command. 3. Obtain users registered. 4. Update a BdD. This is possible? There are any best way to implement this? Thanks a lot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM400P: Sharing IRQS?
On Wed, 2004-06-02 at 04:27, Tony Mountifield wrote: In article [EMAIL PROTECTED], Leo Ann Boon [EMAIL PROTECTED] wrote: The new TDM400P with FXO doesn't take up any IRQ. I've 2 boards and both are not using any IRQ. Weird - does that mean they can't provide Zaptel timing like the X100P can? The TDM400P takes one IRQ. The modules for the TDM400P do not take up any additional IRQs -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simultaneous ring internal extension and external phone number?
On Wed, 2004-06-02 at 01:01, Tracy R Reed wrote: Note however that this WILL NOT work if one of the devices you are calling is on a Zap channel. I have a PRI and I would love to ring my cell phone AND my desk phone (SIP) at the same time but if I try only the Zap interface rings. I posted regarding this a few days ago. It seems silly to have to go out through another VOIP provider when I have my own PRI. I have clients who want this feature too so I would really like to solve this problem. This is a problem with ANALOG interfaces, but not normally an issue with PRI (aka DIGITAL) interfaces. Something else is going on that's causing this problem for you. I can see callprogress= or busydetect= causing something like this. These two options are designed for analog interfaces and I don't know what sorts of issues would happen if you tried using them with PRIs. One of the biggest reasons people use PRIs is so they don't have this problem. I don't know what to suggest to you, other than not to give up. This is a fixable issue with PRIs. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Controlling SIP mobile extensions.
Hi, XISCOAIR wrote: Hi everybody, I'm trying to develop a web application for controlling if SIP users are registered in * or not, and show it in a web. My problem is that I don't now if it's possible to do a Shell Script to control this: 1. Connect to console. 2. Execute command. 3. Obtain users registered. 4. Update a BdD. This is possible? There are any best way to implement this? Thanks a lot. It can be done, in fact it's already done. Look here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20GUI Monastery does exactly what you describe and a bit more. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware)
It seem there are trouble with some sound file about checksum calculation By example I have a wav file = 99kb after converted in ul = 39 kb , but makering give me checksum error !! I trying a wav file recorded with voice recorder, work fine , just chunk error message checksum before = db8e checksum after = 4db2 checksum failed Olivier. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Shaun Ewing Envoyé : mercredi 2 juin 2004 01:46 À : [EMAIL PROTECTED] Objet : RE: [Asterisk-Users] Re: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: Wednesday, 2 June 2004 5:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware) Now if I could only get my GS phones to load the ring tone files. The TFTP log shows all the requests for the usual boot files and the cfg files but NO requests for the ring tones, not even file not found responses. I can't believe that this is the tftp server. I have tried it on at least three different phones, purchased in 2 different lots and still no luck. Maybe the phones just don't like me. Maybe :-) I successfully converted some of the ringtones I had for our 7940/7960 phones which were loaded onto the GS phones with no problems. I just spotted the response from Tony Mountifield - if that's the case, make sure they're not bigger than 65536 bytes. Ours are all around the 3 bytes mark. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
I have this two ip at the same machine, but I tried it using the both address, the result is the same. Kind regards, Miguel Date: Wed, 02 Jun 2004 13:50:05 +1000 From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Firefly version Reply-To: [EMAIL PROTECTED] the log looks legit except why does asterisk have a different IP in the contact compared to the 'to' address. I can connect successfully to my asterisk server and FWD - can anyone give me sip access to a asterisk server that firefly doesn't work on? [EMAIL PROTECTED] wrote: Why all the time the firefly show me the message: Sip registration failed for the network Home (407). The server, username and password are correct. I'm using the default RTP port 5000 in the SIP tab. Using the SJPhone I can register; using the firefly I can call any registered number, but I can't register. On asterisk console: Sip read: REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 User-Agent: Firefly Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.199.121:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 SAMPLANET1*CLI Sip read: REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 Proxy-Authorization: Digest username=2003,realm=asterisk,nonce=38165263,uri=sip:192.168.199.3 :5060; transport=udp,response=ec0afc0a2b13a725aa40b5c311c396d8,algorithm=M D5 User-Agent: Firefly Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.199.121:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with TDM400P
On Wed, 2004-06-02 at 00:33, Wim Kerkhoff wrote: Following problems have been observed, and are preventing us from dumping our existing Nortel Merdian PBX: 1. echo at beginning of call for several seconds, even with various combinations of echocancel and echotraining in zapata.conf Echo is miserable to try to fix. Newer zaptel CVS checkouts have a tool called zttest What are the results of running zttest? Unbalanced lines can cause echo, both IRQ sharing, IDE DMA, framebuffer, and graphics (as well as crappy motherboards) can introduce latency on the PCI bus and cause echo. 2. even though multiple incoming lines are connected, only the first ZAP channel is picking up. So if one line is in use, nobody else can call in even though there are other lines free. When in debug mode (-gcvvv) nothing is showing up that there's another call coming in. This is not a general problem. It sounds like you are not using group= in your zapata.conf to put the TDM ports into a hunt group. 3. channels don't always hang up properly - HookState shows as offhook for quite some time. Sounds like something isn't providing far end disconnect supervision. 4. Asterisk Zap channels don't see an incoming call until 2 rings after the existing Nortel PBX sees it. Both people calling in and people answering don't like that. You don't have Caller*ID on your lines, but Asterisk is configured to use (and waiting for) Caller*ID. See the usecallerid= and callerid= options. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNS SRV records
My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer with Budgetone
Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person: 1. call to me 2. Hold the call and call to other person. 3. I say Anyone want talk to you, OK, thanks, 4. I hangup and first person is directly redirect to second person? It is possible with asterisk and budgetone phones? Regards, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simultaneous ring internal extension and external phone number?
Eric Wieling wrote: On Wed, 2004-06-02 at 01:01, Tracy R Reed wrote: Note however that this WILL NOT work if one of the devices you are calling is on a Zap channel. I have a PRI and I would love to ring my cell phone AND my desk phone (SIP) at the same time but if I try only the Zap interface rings. I posted regarding this a few days ago. It seems silly to have to go out through another VOIP provider when I have my own PRI. I have clients who want this feature too so I would really like to solve this problem. This is a problem with ANALOG interfaces, but not normally an issue with PRI (aka DIGITAL) interfaces. Something else is going on that's causing this problem for you. I can see callprogress= or busydetect= causing something like this. These two options are designed for analog interfaces and I don't know what sorts of issues would happen if you tried using them with PRIs. One of the biggest reasons people use PRIs is so they don't have this problem. I don't know what to suggest to you, other than not to give up. This is a fixable issue with PRIs. I don't have any issues at all doing this with PRI. As a matter of fact, I'm ringing two SIP phones, an IAX phone and three PSTN phones (via a PRI) at the same time. Whoever answers first gets the call. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] isdn configuration
Hi, I have installed Asterisk with sip clients and an ISDN card from Billion. From an ISDN phone I can dial the Asterisk and hear the welcome message, hear the echo test etc. I want to use Asterisk as a gateway between PSTN and SIP so that callers to my ISDN will be transferred to my fwd account and/or the SIP clients connected to Asterisk. I assume my modem.conf is configured correctly, as long as I can call Asterisk from ISDN. That leaves extensions.conf, but I am not sure how to go about. Does anyone have some example configuration to share? Thor -- Using Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Multi process of *
Matteo Brancaleoni [EMAIL PROTECTED] wrote: [...] User Mode Linux is way better for that use, much more efficient. VoIP-only Asterisk also works nicely under vservers (see www.linux-vserver.org), which is even more efficient than UML. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] New Firefly version
Please, send to me. Kind regards, Miguel Date: 2 Jun 2004 04:39:39 - From: muralikrishnan lakshmanan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: RE: [Asterisk-Users] New Firefly version Reply-To: [EMAIL PROTECTED] This is a multipart mime message --Next_1086151179---0-202.54.124.130-19795 Content-type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Content-Disposition: inline P=0Anbsp; BR=0Au hv to change ur sip.conf amp; extensions.conf file i= f u want i will send u.BR=0ABR=0ABR=0AOn Tue, 01 Jun 2004 Paul Mahler= wrote :BR=0Agt;I'm having this problem too.BR=0Agt;BR=0Agt;BR= =0Agt;Paul MahlerBR=0Agt;[EMAIL PROTECTED]BR=0Agt;Signate, LLCBR= =0Agt;665 Third StreetBR=0Agt;Suite 100BR=0Agt;San Francisco, CABR= =0Agt;nbsp; 94107-1901BR=0Agt;BR=0Agt;nbsp; Asterisk Services and = TrainingBR=0Agt;BR=0Agt;BR=0Agt;BR=0Agt;BR=0Agt;BR=0Agt;= BR=0Agt;BR=0Agt;BR=0Agt;BR=0Agt; gt; -Original Message-= BR=0Agt; gt; From: [EMAIL PROTECTED]BR=0Agt; gt= ; [mailto:[EMAIL PROTECTED] On Behalf OfBR=0Agt; g= t; [EMAIL PROTECTED]BR=0Agt; gt; Sent: Tuesday, June 01, 2004 7:53= AMBR=0Agt; gt; To: [EMAIL PROTECTED]BR=0Agt; gt; Sub= ject: [Asterisk-Users] New Firefly versionBR=0Agt; gt;BR=0Agt; gt; = Why all the time the firefly show me the message: SipBR=0Agt; gt; regis= tration failed for the network Home (407).BR=0Agt; gt;BR=0Agt; gt; = The server, username and password are correct. I'm using theBR=0Agt; gt= ; default RTP port 5000 in the SIP tab.BR=0Agt; gt;BR=0Agt; gt; Usi= ng the SJPhone I can register; using the firefly I canBR=0Agt; gt; call= any registered number, but I can't register.BR=0Agt; gt;BR=0Agt; g= t; On asterisk console:BR=0Agt; gt;BR=0Agt; gt; Sip read:BR=0Agt= ; gt; REGISTER sip:192.168.199.3:5060;transport=3Dudp SIP/2.0BR=0Agt; = gt; To: lt;sip:[EMAIL PROTECTED]:5060;transport=3Dudpgt;BR=0Agt; gt;= From: lt;sip:[EMAIL PROTECTED]:5060gt;;tag=3D5a1c4f36BR=0Agt; gt;= Via: SIP/2.0/UDPBR=0Agt; gt; 192.168.199.121:5060;branch=3Dz9hG4bK-c87= 542-436999556-1--c87542-;rportBR=0Agt; gt; Call-ID: c90fa011e82acf3eBR= =0Agt; gt; CSeq: 1 REGISTERBR=0Agt; gt; Contact: lt;sip:[EMAIL PROTECTED] 8.199.121:5060gt;BR=0Agt; gt; Expires: 3600BR=0Agt; gt; Max-Forwar= ds: 70BR=0Agt; gt; User-Agent: FireflyBR=0Agt; gt; Content-Length: = 0BR=0Agt; gt;BR=0Agt; gt;BR=0Agt; gt; 11 headers, 0 linesBR= =0Agt; gt; Using latest request as basis requestBR=0Agt; gt; Sending = to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT):BR=0Agt; gt; = SIP/2.0 100 TryingBR=0Agt; gt; Via: SIP/2.0/UDPBR=0Agt; gt; 192.168= .199.121:5060;branch=3Dz9hG4bK-c87542-436999556-1--c87542-;rportBR=0Agt;= gt; From: lt;sip:[EMAIL PROTECTED]:5060gt;;tag=3D5a1c4f36BR=0Agt;= gt; To: lt;sip:[EMAIL PROTECTED]:5060;transport=3Dudpgt;;tag=3Das7843e= 908BR=0Agt; gt; Call-ID: c90fa011e82acf3eBR=0Agt; gt; CSeq: 1 REGIS= TERBR=0Agt; gt; User-Agent: Asterisk PBXBR=0Agt; gt; Allow: INVITE,= ACK, CANCEL, OPTIONS, BYE, REFERBR=0Agt; gt; Contact: lt;sip:[EMAIL PROTECTED] .168.199.4gt;BR=0Agt; gt; Content-Length: 0BR=0Agt; gt;BR=0Agt;= gt;BR=0Agt; gt;nbsp; to 192.168.199.121:5060BR=0Agt; gt; Transmi= tting (no NAT):BR=0Agt; gt; SIP/2.0 407 Proxy Authentication RequiredB= R=0Agt; gt; Via: SIP/2.0/UDPBR=0Agt; gt; 192.168.199.121:5060;branch= =3Dz9hG4bK-c87542-436999556-1--c87542-;rportBR=0Agt; gt; From: lt;sip:= [EMAIL PROTECTED]:5060gt;;tag=3D5a1c4f36BR=0Agt; gt; To: lt;sip:20= [EMAIL PROTECTED]:5060;transport=3Dudpgt;;tag=3Das7843e908BR=0Agt; gt; = Call-ID: c90fa011e82acf3eBR=0Agt; gt; CSeq: 1 REGISTERBR=0Agt; gt; = User-Agent: Asterisk PBXBR=0Agt; gt; Allow: INVITE, ACK, CANCEL, OPTION= S, BYE, REFERBR=0Agt; gt; Contact: lt;sip:[EMAIL PROTECTED]gt;BR= =0Agt; gt; Proxy-Authenticate: Digest realm=3Dquot;asteriskquot;, nonce= =3Dquot;38165263quot;BR=0Agt; gt; Content-Length: 0BR=0Agt; gt;B= R=0Agt; gt;BR=0Agt; gt;nbsp; to 192.168.199.121:5060BR=0Agt; gt= ; SAMPLANET1*CLIgt;BR=0Agt; gt;BR=0Agt; gt; Sip read:BR=0Agt; = gt; REGISTER sip:192.168.199.3:5060;transport=3Dudp SIP/2.0BR=0Agt; gt;= To: lt;sip:[EMAIL PROTECTED]:5060;transport=3Dudpgt;BR=0Agt; gt; Fr= om: lt;sip:[EMAIL PROTECTED]:5060gt;;tag=3D6c3de14aBR=0Agt; gt; Vi= a: SIP/2.0/UDPBR=0Agt; gt; 192.168.199.121:5060;branch=3Dz9hG4bK-c87542= -373911025-1--c87542-;rportBR=0Agt; gt; Call-ID: c90fa011e82acf3eBR= =0Agt; gt; CSeq: 1 REGISTERBR=0Agt; gt; Contact: lt;sip:[EMAIL PROTECTED] .199.121:5060gt;BR=0Agt; gt; Expires: 3600BR=0Agt; gt; Max-Forward= s: 70BR=0Agt; gt; Proxy-Authorization: DigestBR=0Agt; gt; username= =3D2003,realm=3Dquot;asteriskquot;,nonce=3Dquot;38165263quot;,uri=3Dqu= ot;sip:192.1BR=0Agt; gt; 68.199.3:5060;BR=0Agt; gt; transport=3Dudp= quot;,response=3Dquot;ec0afc0a2b13a725aa40b5c311c396d8quot;,algBR=0Ag= t; gt; orithm=3DMD5BR=0Agt; gt; User-Agent: FireflyBR=0Agt; gt; Co= ntent-Length: 0BR=0Agt; gt;BR=0Agt; gt;BR=0Agt; gt; 12 headers,= 0 linesBR=0Agt; gt; Using latest request as basis requestBR=0Agt; = gt; Sending to 192.168.199.121 :
Re: [Asterisk-Users] Sipura-SPA2000 background noise
I too have the same problem on a few units, but not on others. I also have been having difficulty hooking up multiple lines from one Sipura to the same multi-line phone system (seems to create a line cross) but have no problems with either cisco or dlink boxes. In general they are nice units, but I suspect they may have had a batch go out that were noisy. -Steve On Jun 1, 2004, at 3:10 PM, [EMAIL PROTECTED] wrote: I hear the exact same noise on 2 units I purchased a few months ago. I've been in contact with sipura support and they are willing to try RMA'ing one of my units. As soon as I can get to the site with the sipura, I'll be sending it in. I'll post my results to the list. btw, I'd have to agree that its not comfort noise, its very similar (only much louder) to the hiss that the old digium fxs modules had on the tdm boards. Mark At 10:21 AM 6/1/2004, you wrote: Not really a comfort noise. I say anything and it doesent go away. It sounds like a shielding issue. I have tried to relocate the unit but it doesn't seem to help. -Original Message- From: Kevin Walsh [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 11:46 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sipura-SPA2000 background noise Kevin [EMAIL PROTECTED] wrote: I have been using Cisco ATA's for analog connections and decided to give a Sipura SPA-2000 a try. I noticed there is a fair amount of background white noise that is noticeable, especially after breaking the dial tone. After pressing a '1' to break the dial tone, there is a fair amount of noise that is evident. I do not notice this condition on the Cisco ATA's. I plugged the Sipura in the same location as the Cisco ATA. Anyone else have this condition with the Sipura? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
On Wednesday 26 May 2004 19:42, Jon Lawrence wrote: It looks like my missing digit problems are down to the dect phone I have connected to my handytone ata-286. When i have my Binatone dect connected, I only get the first 8 digits, if I connect my panasonic dect then I see all the digits - looks like I need a different dect phone :( Any ways, It looks like the patch works perfectly to me. It also works fine on my Telewest (Eurobell). I'm even more confused now. If I have the number in the phones phone book then it will show the relevant name, otherwise it only shows the first 8 digits. Has anyone ever heard of anything like this ? TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware)
Stig Hess wrote: Now if I could only get my GS phones to load the ring tone files. The TFTP log shows all the requests for the usual boot files and the cfg files but NO requests for the ring tones, not even file not found responses. I can't believe that this is the tftp server. I have tried it on at least three different phones, purchased in 2 different lots and still no luck. Maybe the phones just don't like me. I have exactly the same problem. Could there be different hardware versions? Stig Can't say I haven't wondered that myself. Given that there are no serial numbers visible on the phones, I suspect that one could use the MAC address as a serial number, since these are probably allocated sequentially. Only the last 6 digits should change. My first phone (bought as an evaluator) has last 6 digits of 002175. The second shipment has numbers in the range of 002935 and up. It would be interesting to know if the MAC's from the phones that work are significantly higher than these. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware)
Tony Mountifield wrote: Make sure the ring tone files are no bigger than 65536 bytes. Earlier versions of my program didn't check for this, but the latest one does. That's potentially important information, but even still, the tftp log would show the phone requesting the file, even if it rejects it later for being overly long. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura-SPA2000 background noise
Hi Brian, Brian Cuthie wrote: BTW, anyone know how to get the SPA-2000 do drop loop current momentarily when the other end hangs up? -brian There is a web configuration option to reverse the polarity in the latest 2.0 firmware. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer with Budgetone
On Wed, 2004-06-02 at 07:41, Sergio Serrano wrote: Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person: 1. call to me 2. Hold the call and call to other person. 3. I say Anyone want talk to you, OK, thanks, 4. I hangup and first person is directly redirect to second person? It is possible with asterisk and budgetone phones? Not that I am aware of -- Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS SRV records
I have the same problem, I have a domain name but I do not want to pay for DNS services... I kept on trying to find a place where I can get SRV records from but none of the free DNS services provide them. I've tried ZoneEdit, DNS Park, etc. I've seen one which there might be a possibility, www.granitecanyon.com.. though they do not offer mail forwarding. Andrew Thompson wrote: My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Adtran TSU 600
Bartosz Jozwiak wrote: Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B. Yes, but it was a while ago (last August). I currently have the TSU600 with 2 FXO/1 FXS cards running on a T100P with the only problem being that the FXS card is a little flakey, but this has no bearing on the T100P. I just downloaded the firmware and the stripped down version of T-Flash that Adtran provides for flashing the firmware. Once the serial link to the TSU was up and running properly, the firmware update went without a hitch. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS SRV records
Andrew Thompson wrote: Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? As with email you technically don't need MX records, an A record will also work fine. I'm pretty sure (long day can't be bothered checking) Asterisk won't even do SRV lookups by default as it's commented out in the default config... In other words you'd be better off with an A record... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Controlling SIP mobile extensions.
XISCOAIR wrote: Hi everybody, I'm trying to develop a web application for controlling if SIP users are registered in * or not, and show it in a web. My problem is that I don't now if it's possible to do a Shell Script to control this: 1. Connect to console. 2. Execute command. 3. Obtain users registered. 4. Update a BdD. This is possible? There are any best way to implement this? The registration is stored in the Asterisk database. It's a db1 database you can read from a perl script directly, without using the manager API. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS SRV records
On Wed, 2004-06-02 at 13:40, Andrew Thompson wrote: My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? Sorry, really needs to be SRV :/ F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Transfer with Budgetone
Sergio Serrano wrote: Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person: 1. call to me 2. Hold the call and call to other person. 3. I say Anyone want talk to you, OK, thanks, 4. I hangup and first person is directly redirect to second person? It is possible with asterisk and budgetone phones? Sergio, Not as far as I know, at least not exactly the way you have outlined it. Try this: 1. call comes to you 2. You hold the call and call other person. 3. You say Someone wants to talk to you, OK, thanks 3a. Other person then hangs up. 3b. You flash back to the original caller 3c. You tell them that you are transferring the call 3d. You transfer the call using the transfer feature on the phone 4. You hangup and first person is transferred to other person? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Multi process of *
User Mode Linux is way better for that use, much more efficient. Matteo. I am using user mode Linux very successfully to run as many asterisks as I need. Besides asterisk, UML is my other favourite open source project with which I am involved developing complete turn key solutions (including much wanted asterisk web interface). If any of you guys do not know or simply do not have time learning how to configure UML, please contact me off the list for UML Web interface. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura-SPA2000 background noise
Nicolas Gudino wrote: Hi Brian, Brian Cuthie wrote: BTW, anyone know how to get the SPA-2000 do drop loop current momentarily when the other end hangs up? -brian There is a web configuration option to reverse the polarity in the latest 2.0 firmware. Yeah, I saw that too. But it doesn't always seem to fire when I think it should. And, my Nortel switch ignores it anyway, since they have conveniently made their trunks polarity insensitive. What would be better is if it dropped loop current entirely for a few hundred milliseconds. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem compiling ZAPTEL on Linux 2.6.6
Hi, I have Debian Linux with kernel 2.6.6. The all packages compiled except ZAPTEL where I have the following error: voipgw:/usr/src/zaptel# make make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.6' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1165: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1237: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: warning: type defaults to `int' in declaration of `__mptr' /usr/src/zaptel/zaptel.c:1293: warning: initialization from incompatible pointer type /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: invalid type argument of `-' /usr/src/zaptel/zaptel.c:1353: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1486: warning: passing arg 1 of `unregister_hdlc_devicevoipgw:/usr/src/zaptel# make make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.6' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1165: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1237: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: warning: type defaults to `int' in declaration of `__mptr' /usr/src/zaptel/zaptel.c:1293: warning: initialization from incompatible pointer type /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: invalid type argument of `-' /usr/src/zaptel/zaptel.c:1353: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1486: warning: passing arg 1 of `unregister_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel/zaptel.c:2950: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:2955: warning: passing arg 1 of `unregister_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c:3035: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3037: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3038: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3040: warning: assignment from incompatible pointer type /usr/src/zaptel/zaptel.c:3047: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3048: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3049: warning: passing arg 1 of `register_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `__zt_getbuf_chunk': /usr/src/zaptel/zaptel.c:4626: structure has no member named `netdev' /usr/src/zaptel/zaptel.c: In function `__zt_putbuf_chunk': /usr/src/zaptel/zaptel.c:5499: structure has no member named `netdev' make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.6' make: *** [linux26] Error 2 Could you be so kind to give me some suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Controlling SIP mobile extensions.
Hi! This is possible? There are any best way to implement this? Yes, look at asterisk -rx command That command then can be sip show peers or database show sip. Here is an example of a related CRON job that I use for restart: # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS SRV records
On Wed, Jun 02, 2004 at 03:14:58PM +0200, Stephen Rosebush said: I have the same problem, I have a domain name but I do not want to pay for DNS services... I kept on trying to find a place where I can get SRV records from but none of the free DNS services provide them. I've tried ZoneEdit, DNS Park, etc. I've seen one which there might be a possibility, www.granitecanyon.com.. though they do not offer mail forwarding. Setup your own master that you manage, and have zoneedit slave. They support this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DNS SRV records
www.xname.org =) -Original Message- From: Stephen Rosebush [mailto:[EMAIL PROTECTED] Sent: 02 June 2004 2:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DNS SRV records I have the same problem, I have a domain name but I do not want to pay for DNS services... I kept on trying to find a place where I can get SRV records from but none of the free DNS services provide them. I've tried ZoneEdit, DNS Park, etc. I've seen one which there might be a possibility, www.granitecanyon.com.. though they do not offer mail forwarding. Andrew Thompson wrote: My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Adtran TSU 600
I just did it today successfully. :) The one BIG problem I have is: there is no battery when I pick up a phone connected to FXS port. My adtran tsu 600 has 24 fxs ports. Please could you tell me what kind of configuration you have in your adtran for fxs ports and what kind of configuration you have for zaptel and zapata. I have been struggling with adtran and T100p for 2 weeks now with no success so far. If it is no problem I could use your help. Thank you in advance. bartosz - Original Message - From: Stephen R. Besch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 02, 2004 10:17 AM Subject: [Asterisk-Users] Re: Adtran TSU 600 Bartosz Jozwiak wrote: Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B. Yes, but it was a while ago (last August). I currently have the TSU600 with 2 FXO/1 FXS cards running on a T100P with the only problem being that the FXS card is a little flakey, but this has no bearing on the T100P. I just downloaded the firmware and the stripped down version of T-Flash that Adtran provides for flashing the firmware. Once the serial link to the TSU was up and running properly, the firmware update went without a hitch. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Splicing audio clips into one stream
Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Transfer with Budgetone
I know that way, but some person ask for me for first way to do transfers. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Stephen R. Besch Enviado el: miércoles, 02 de junio de 2004 15:37 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Re: Transfer with Budgetone Sergio Serrano wrote: Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person: 1. call to me 2. Hold the call and call to other person. 3. I say Anyone want talk to you, OK, thanks, 4. I hangup and first person is directly redirect to second person? It is possible with asterisk and budgetone phones? Sergio, Not as far as I know, at least not exactly the way you have outlined it. Try this: 1. call comes to you 2. You hold the call and call other person. 3. You say Someone wants to talk to you, OK, thanks 3a. Other person then hangs up. 3b. You flash back to the original caller 3c. You tell them that you are transferring the call 3d. You transfer the call using the transfer feature on the phone 4. You hangup and first person is transferred to other person? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some (lack of) answers regarding the wakeup call application...
I installed your wakeup agi script and it works well. There are some typos on the wiki page--the printf format string seem to have been corrupted. Also, I need a Linux tool to splice a series of gsm audio clips together in order to use one 'get_data' instead of multiple 'stream_file' commands. Mike Rob Fugina wrote: Since I only seem to get questions, and no feedback, from the Wiki page, I'll ask here. There seems to be no lack of opinions here... I have a working wakeup call system on my home * system. The architecture is something I'm not perfectly happy with, though. There are two AGI scripts, written in Perl, which handle (a) scheduling, confirming, and cancelling a wakeup call, and (b) the wakeup call itself, with the option to snooze for 5, 15, or 30 minutes. The Perl scripts use the Asterisk::AGI module I came across months ago, but by necessity, can't use the Asterisk/Perl code for creating call files -- it has a habit of creating them right in the outgoing call queue, generating a call immediately. So the Perl code creates call files in a wakeup queue directory, and a cron job (a shell script) runs every minute looking for wakeup calls in the queue that need to be handled, and moves them to the outgoing call queue. It has occurred to me that the two AGI scripts could be rewritten as real compiled asterisk applications, but then it always hits me that without some kind of cron-line built-in scheduler, or changes to the outgoing call queueing that would allow a call to be scheduled for the future, there would still be that external cron-driven shell script. Ugly. What I'm wondering is this: Is there enough interest in the new features I mentioned (either an internal scheduler or scheduled outgoing calls) that I should work on a C version of the wakeup AGI scripts, or should my (impending) next rewrite maintain the current architecture? Anyone with specific questions about using my wakeup app, please email me directly. Rob -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware)
In article [EMAIL PROTECTED], vv [EMAIL PROTECTED] wrote: It seem there are trouble with some sound file about checksum calculation By example I have a wav file = 99kb after converted in ul = 39 kb , but makering give me checksum error !! I trying a wav file recorded with voice recorder, work fine , just chunk error message checksum before = db8e checksum after = 4db2 checksum failed Are you running the perl program on Unix/Linux or on Windows? It has only been tested on Linux, and may need binmode STDIN; if running under Windows. Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FireFly - no sound after first call
I've been watching to see if this problem comes up with anyone elses firefly - but so far i'm the only one experiencing the problem. When I connect to either my asterisk server or FWD all goes well on the first call. I can hear and talk. But every call after the first one I end up with no sound - not even ringing. I use win98 and have tried it on two systems with win98 installed. thanks joe p.s. if any other info can provide let me know. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Sip/IP Phones
Hi there, I want to buy a IP Phone but i found it rather ro ask the asterisk mailinglist... Does anybody uses a Grandstream 1XX and have probles with the asterisk? Wich phone would you me rate? in a price range from 100 - 150$ ? Best Regards, Mark Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Transfer with Budgetone
Stephen R. Besch wrote: Not as far as I know, at least not exactly the way you have outlined it. Try this: 1. call comes to you 2. You hold the call and call other person. 3. You say Someone wants to talk to you, OK, thanks 3a. Other person then hangs up. 3b. You flash back to the original caller 3c. You tell them that you are transferring the call 3d. You transfer the call using the transfer feature on the phone 4. You hangup and first person is transferred to other person? Ugh. So Asterisk doesn't handle transfer? Every company phone system I've ever used has not required 3a-3d. It looks like a real hack to do so. It anyone working on implementing this? Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails
Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying since it always leaves an empty VM. thanks Dominique ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme with moderator
All, I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way? Bruce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk process respawn
Anyone know how to place asterisk in initab so that it is loaded at boot and will respawn if the process goes down? I
Re: [Asterisk-Users] DNS SRV records
Duane wrote: Andrew Thompson wrote: Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? As with email you technically don't need MX records, an A record will also work fine. I'm pretty sure (long day can't be bothered checking) Asterisk won't even do SRV lookups by default as it's commented out in the default config... In other words you'd be better off with an A record... Spoken like a true n00b13. You can *sometimes* get away with not having MX records. You can *sometimes* get away with not having SVR records. Both record types exist for a reason though. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Splicing audio clips into one stream
Michael Welter wrote: Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Thanks, cat clip1.gsm newclip.gsm ; cat clip2.gsm newclip.gsm The pipe means append to the end of the file. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Splicing audio clips into one stream
On Wed, 2004-06-02 at 09:22, Michael Welter wrote: Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Think for a bit more about how AGI works. basically you will want to stream n number of audio clips, but if any one of them gets interupted, you want to pause and listen for the input from user. Not a problem. Once you detect input other than prompt complete, you keep listening(wait for digit) till you have enough data to do something. If splicing clips together was what was needed, you would see a built in tool to do such a thing. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: determining cause of dropped calls?
I am having a similar problem. It is not frequent, perhaps once in 80-100 calls. CVS-HEAD-05/08/04-21:57:50 using Cisco 7960 6.3 and X100P --__--__-- Date: Tue, 1 Jun 2004 21:04:14 -0700 (PDT) From: Bruce Komito [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] determining cause of dropped calls? I am trying to figure out why calls between SIP devices and the PSTN are being regularly dropped after anywhere from 2-15 minutes. I have turned on everything I can think of, but I don't see any obvious reasons for the drops. All I can see from turning on debug and verbosity is two messages advising of a destroyed call, followed by normal-looking SIP and ZAP termination messages. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_rtp_read: Unknown RTP codec
Any one see these? Are they benign, or is some system tuning required to remove them? Can't seem to find a resolution in the archives. If you have a link, it would be appreciated. Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Jun 2 10:58:59 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:00 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:01 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem compiling ZAPTEL on Linux 2.6.6
On Wed, 2004-06-02 at 14:56, Miroslav Nachev wrote: I have Debian Linux with kernel 2.6.6. The all packages compiled except ZAPTEL where I have the following error: voipgw:/usr/src/zaptel# make make linux26 F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Splicing audio clips into one stream
cat 1.gsm 2.gsm 3.gsm new.gsm works fine James On Wed, 2 Jun 2004, Michael Welter wrote: Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P: Sharing IRQS?
On 1-Jun-04, at 6:57 PM, Leo Ann Boon wrote: The new TDM400P with FXO doesn't take up any IRQ. I've 2 boards and both are not using any IRQ. Can you please double-check? I have 2 servers, each with a tdm400p + quad-fxo ... with both of these installs, the card is assigned an interrupt: # more /proc/interrupts CPU0 0: 16400284 XT-PIC timer 1: 5 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 10: 163940440 XT-PIC wctdmTDM400P + quad-fxo 12: 643996 XT-PIC eth0, PS/2 Mouse 14: 165056 XT-PIC ide0 15: 34 XT-PIC ide1 Thanks Ryan Isamar Maia wrote: I had a little nightmare playing with X100Ps and IRQs and I decided to buy TDMP400P/FXO and FXS. The question is, can I put multiple boards in the same motherboard without worrying about IRQS? TDM400P shares IRQs with other boards? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ryan Courtnage Coalescent Systems 403.244.8089 www.voxbox.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Splicing audio clips into one stream
On Wed, 02 Jun 2004 08:22:23 -0600, Michael Welter [EMAIL PROTECTED] wrote: Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. You could use: cat file1 file2 file3 bigfile but it wouldn't have any pause in between the files. -- -S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Transfer with Budgetone
On Wed, 2004-06-02 at 09:44, Tony Hoyle wrote: Ugh. So Asterisk doesn't handle transfer? Every company phone system I've ever used has not required 3a-3d. It looks like a real hack to do so. It anyone working on implementing this? As far as I can tell it's a limitation of the phone, not of Asterisk. Most phones seem to implement the type of transfer you are wanting to do as a special form of a 3-way call. The phone you have doesn't support 3-way calls as documented on: http://www.grandstream.com/Product_Spec.pdf Other IP phones like the Cisco DO support 3-way calling and support supervised/consultative transfers (which is the term for what you want to do) -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and multiple line appearances
I am learning about SIP phones little by little. I've been working with * for about 2 weeks now with FXS and FXO ports and analog phones, but we want to evaluate the utility of going to SIP phones directly from here rather than investing in analog phones first. One of the questions I have is multiple line appearance on phones. Since the phone basically acts like a computer, there is no reason that any SIP phone couldn't accommodate call waiting, multiple lines, etc. And the info I've found on the net indicates that SIP phones *can* handle multiple line appearances. My question is whether it is the normal case that SIP phones would do such a thing. e.g. We are looking at various phones (Snom is one), but none of their phones mention multiple line appearances. I don't know if this is because they cannot handle it or because it's an assumed feature of SIP phones. -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk process respawn
ax:2:respawn:/usr/sbin/asterisk -vvvcgf Nate Turnbow Systems Engineer CHG Companies On Wed, 02 Jun 2004 10:01:34 -0500 Terry Goodwin [EMAIL PROTECTED] wrote: Anyone know how to place asterisk in initab so that it is loaded at boot and will respawn if the process goes down? I -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme with moderator
Hi Bruce, I am doing smth similar in one appli. just using MeetMecount inside an AGI script, btw I know when there is someone in the conference and I can let the user go in or not! Thr trick is to save the result in a variable and then use GET VARIABLE to get the nb user! / snprintf( tmp_command, 200, EXEC MeetMeCount %d|count, room_number); ... res = run_command(GET VARIABLE count); -/ I can give you more input if you need, I didnt want to past a lot of source code in the mailing-list. Hope it helps, Kind regards, Areski On Wed, 2004-06-02 at 16:57, Bruce Marler wrote: All, I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way? Bruce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Feature request for integrating an OSS (Operations Support System) and Asterisk
Hi, I work for an ISP/CLEC, and we have developed our own OSS (Operations Support System), which handles all billing, sales, provisioning, and support issues. When it was originally being designed, the idea was to integrate it with Asterisk. Other than Caller-ID information (so that past trouble tickets, and billing issues can be brought up for the agent), how else would the Asterisk community like the OSS we developed and Asterisk to interact (perhaps transferring calls, etc.)? More information about the OSS is here: http://www.vylink.com/oss/ Also, if you have any other suggestions for features that aren't on the webpage, feel free to email [EMAIL PROTECTED] If there is enough demand for a feature we don't have or we like the feature enough, we will likely add it. Thanks, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax Recognizion without Answer? How to Supress this?
Hi, I have really googled and read the wiki but I still no idea, how to supress the fax recognizion. Our users are not able to fax and that is bad... Could you give me an hint, please? Thanks Felix Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri * -- Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are recognized as faxes and * tries to forward the call to fax. I do not answer this calls... == Spawn extension (Amt595xxx-In, 595164, 1) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' -- Starting simple switch on 'Zap/62-1' -- Accepting overlap call from '595457' to '034491' on channel 31, span 2 -- Executing SetVar(Zap/62-1, Out=Zap/g1/) in new stack -- Executing Goto(Zap/62-1, OutDial-LCR|BYEXTENSION|1) in new stack -- Goto (OutDial-LCR,034491***,1) -- Executing SetVar(Zap/62-1, LCR=01081) in new stack -- Executing Goto(Zap/62-1, OutDial-Dial|BYEXTENSION|1) in new stack -- Goto (OutDial-Dial,034491,1) -- Executing Dial(Zap/62-1, Zap/g1/0108103|30|TrH) in new stack -- Called g1/010810344918*** -- Redirecting Zap/62-1 to fax extension -- Hungup 'Zap/1-1' == Spawn extension (OutDial-Dial, fax, 0) exited non-zero on 'Zap/62-1' -- Executing Dial(Zap/62-1, Zap/g1/01081fax|30|TrH) in new stack -- Called g1/01081fax -- Channel 2, span 1 got hangup -- Hungup 'Zap/2-1' What have I to change? Could I supress that? Thanks Felix Deierlein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 403 Forbidden between two softphones on same Asterisk
On Wed, Jun 02, 2004 at 11:25:26AM +0200, Tor Houghton wrote:
Hi,
I have two softphones connected to an Asterisk stable. I have two
extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be
completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on
extension 2000 will ring, but as soon as the call is picked up, extension
2000 will hang up the call.
[snip]
I seem to have resolved this problem; for some reason, when upgrading from
an earlier version, the following line was invalid:
exten = 2000,2,Dial(${PHONE1},20,Ttm)
I replaced it with
exten = 2000,2,Dial(${PHONE1},20,t)
And it works fine. I guess I misunderstood the flags during an earlier
configuration of the extensions. Sorry to bother you all.
Tor
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Re: [Asterisk-Users] Asterisk Receptionist manager program.
I put an update on the website. It fixes the IAX calls crashing the program. ANd added Voice Mail checking. It will tell you how many new and old voice mails are in your box. http://www.easyhomenetworks.com/AstRec/ Kyle John Fraizer wrote: Kyle Hagan wrote: Ok I have a testing version available at www.easyhomenetworks.net/astrec There is a shot docs.txt in the directory you will need to read. Its very very beta (alpha?). There are a couple bugs right now. But give me your ideas and CONSTRUCTIVE critisism please. :) It will only transfer calls for the extention setup in the config file. But will transfer to any extension setup of the astrec.conf including to valetparkedcall addon. Ithas a bug im working on where it wont transfer a Zap call that should be fixed today. But will transfer internal extentions. Im going to create a web site for Asterisk Receptionist soon at www.easyhomenetworks.net/astrec with updates. Kyle Hiya Kyle. I installed the Beta just to play with and found a problem: When a call comes inbound from an IAX2 trunk, I get Runtime Error '5' Invalid procedure call or argument. When you click OK, the app closes. Calling out an IAX2 trunk does the same thing. Calls from SIP to SIP on the same server seem to work fine, as do calls to SIP off the server. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme with moderator
Hi, -Original Message- I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way? Simple extension logic is enough to do this: From a certain extension or with a special pincode or whatever, have moderator access. Be sure to set a database entry (/MMModerator/Roomnr/ = 1) before accessing the MeetMe. For all others, first check this database entry. Only access MeetMe if the flag is set. Something like this ? There are many other ways to achieve this goal. You have to choose the approach that suits you best :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS SRV records
John Fraizer wrote: Spoken like a true n00b13. If the current SIP bug isn't annoying enough to push people away from asterisk you just have to chip in your 2 cents worth to push things that little bit more... You can *sometimes* get away with not having MX records. You can *sometimes* get away with not having SVR records. Both record types exist for a reason though. Oh so that's why SRV lookups are commented out in the default asterisk config, so you can't get anything? sip://[EMAIL PROTECTED] works perfectly well... Before you berate others indescriminately remove your foot from your mouth next time so you don't look like as big of an ass next time... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 and cause code 'user busy'
Hi all, I just installed chan_h323 to interface to a H.323/ISDN gateway. It works really well after two days learning and testing except one thing somebody of you may have an answer to: If I call in from PSTN to a busy asterisk SIP extension I can see a SIP status 486 BUSY, but don't get it passed to the H.323/ISDN side. Asterisk jumps correctly to EXTEN+101 in the dialplan. I tried different Apps there (Hangup, Busy, Congestion) They deliver different cause codes to the H.323/ISDN side (normal call clearing or call rejected) but none of them returns 'user busy' as expected. In Zaptel with Q.931 PRI (euroisdn) you can do exten = 123,102,SetVar(PRI_CAUSE=17) exten = 123,103,Hangup to explicitely set the RELEASE cause code. Is something similiar also possible with H.323? Thank you and regards, Jan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Problem compiling ZAPTEL on Linux 2.6.6
Hello Fran, I try with make linux26 but the result is the same: voipgw:/usr/src/zaptel# make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.6' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1165: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1237: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: warning: type defaults to `int' in declaration of `__mptr' /usr/src/zaptel/zaptel.c:1293: warning: initialization from incompatible pointer type /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: invalid type argument of `-' /usr/src/zaptel/zaptel.c:1353: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1486: warning: passing arg 1 of `unregister_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel/zaptel.c:2950: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:2955: warning: passing arg 1 of `unregister_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c:3035: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3037: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3038: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3040: warning: assignment from incompatible pointer type /usr/src/zaptel/zaptel.c:3047: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3048: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3049: warning: passing arg 1 of `register_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `__zt_getbuf_chunk': /usr/src/zaptel/zaptel.c:4626: structure has no member named `netdev' /usr/src/zaptel/zaptel.c: In function `__zt_putbuf_chunk': /usr/src/zaptel/zaptel.c:5499: structure has no member named `netdev' make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.6' make: *** [linux26] Error 2 -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Wednesday, June 2, 2004, 6:17:12 PM, you wrote: FB On Wed, 2004-06-02 at 14:56, Miroslav Nachev wrote: I have Debian Linux with kernel 2.6.6. The all packages compiled except ZAPTEL where I have the following error: voipgw:/usr/src/zaptel# make FB make linux26 FB F FB ___ FB Asterisk-Users mailing list FB [EMAIL PROTECTED] FB http://lists.digium.com/mailman/listinfo/asterisk-users FB To UNSUBSCRIBE or update options visit: FBhttp://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails
Hi, I've just tried it in my setup and it does not occur anymore. I did see this problem when I first got the phone, but since then I've updated everything, and it appears to have gone away :-) asterisk CVS-04/10/04-15:32:35 ZyXel P 2000 Software version WJ.00.0a bootrom version B.00.13 release date Apr 12 2004 Cheers Giles Scott - Original Message - From: Dominique Kull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 02, 2004 3:46 PM Subject: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying since it always leaves an empty VM. thanks Dominique ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax Recognizion without Answer? How to Supress this?
Either put the channel that your fax machine is on in a context without exten = fax or remove the exten = fax from the context the fax machine is in. The exten = fax is ONLY needed if you want to share an inbound line between fax and voice. On Wed, 2004-06-02 at 11:01, ePyron Felix Deierlein wrote: Hi, I have really googled and read the wiki but I still no idea, how to supress the fax recognizion. Our users are not able to fax and that is bad... Could you give me an hint, please? Thanks Felix Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri * -- Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are recognized as faxes and * tries to forward the call to fax. I do not answer this calls... == Spawn extension (Amt595xxx-In, 595164, 1) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' -- Starting simple switch on 'Zap/62-1' -- Accepting overlap call from '595457' to '034491' on channel 31, span 2 -- Executing SetVar(Zap/62-1, Out=Zap/g1/) in new stack -- Executing Goto(Zap/62-1, OutDial-LCR|BYEXTENSION|1) in new stack -- Goto (OutDial-LCR,034491***,1) -- Executing SetVar(Zap/62-1, LCR=01081) in new stack -- Executing Goto(Zap/62-1, OutDial-Dial|BYEXTENSION|1) in new stack -- Goto (OutDial-Dial,034491,1) -- Executing Dial(Zap/62-1, Zap/g1/0108103|30|TrH) in new stack -- Called g1/010810344918*** -- Redirecting Zap/62-1 to fax extension -- Hungup 'Zap/1-1' == Spawn extension (OutDial-Dial, fax, 0) exited non-zero on 'Zap/62-1' -- Executing Dial(Zap/62-1, Zap/g1/01081fax|30|TrH) in new stack -- Called g1/01081fax -- Channel 2, span 1 got hangup -- Hungup 'Zap/2-1' What have I to change? Could I supress that? Thanks Felix Deierlein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
