Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
On Mon, Mar 3, 2008 at 6:27 AM, Prashant Sharma [EMAIL PROTECTED] wrote: Hi Tilghman, Thanks for taking interest in my problem. I just want to send a http post request to my website without changing the dial plan. So I have added slightly modified http post code and some other code to channel.c got from curl/curl.h. After adding the code I compiled the asterisk code and got the error: channel.o(.text+0x): channel.c:: undefined reference to 'curl_global_init' Try adding the following above the first include in channel.c. It's what I've done in app_dial.c to get the Curl library to link properly. /*** MODULEINFO dependCurl/depend ***/ Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
On Mon, 2008-03-03 at 10:14 +0800, NOC Ph wrote: Hi Guys, I’m new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I’m planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I see it or generate it? Because I have to monitor people who’s calling overseas. 2. How do I secure it? Co’z I have to open it via Public IP. Can I know the port asterisk used assuming I’ll use SIP. 3. If it run on linux, it run will on BSD but I read from google that it has specific version for BSD. Can I know what version are for FreeBSD? Hi, Just a quick one. Last month i created a asterisk system running on freebsd-7 (pre-release) Nothing special, Took an 1.4 asterisk from the ports tree. Only minor point is that it takes foreven and a day to rebuild the whole lot, but that is because i had to use old hardware. (bsd is pick about hardware) But on the whole, it works nicely including the tdm-board. hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
2008/3/3 Hans Witvliet [EMAIL PROTECTED]: On Mon, 2008-03-03 at 10:14 +0800, NOC Ph wrote: Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I see it or generate it? Because I have to monitor people who's calling overseas. 2. How do I secure it? Co'z I have to open it via Public IP. Can I know the port asterisk used assuming I'll use SIP. 3. If it run on linux, it run will on BSD but I read from google that it has specific version for BSD. Can I know what version are for FreeBSD? Hi, Just a quick one. Last month i created a asterisk system running on freebsd-7 (pre-release) Nothing special, Took an 1.4 asterisk from the ports tree. Only minor point is that it takes foreven and a day to rebuild the whole lot, but that is because i had to use old hardware. (bsd is pick about hardware) But on the whole, it works nicely including the tdm-board. hw I am going to chime in for the best turnkey solution. Switchvox. I have setup a couple switchvox boxes (long before they were purchased by Digium) and could not have been happier. Great UIs, functionality, documentation, and support. Thirdlane would be my second choice, followed by FreePBX. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
On Monday 03 March 2008 00:27:32 Prashant Sharma wrote: I just want to send a http post request to my website without changing the dial plan. This doesn't make any sense. The dialplan is among the easiest and least bug-prone ways of adding a curl POST call. Why would you want to code this directly into channel.c, where it does not belong, instead of making a very simple change to your dialplan? Don't explain the mechanics. Explain why you're doing it. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
Hi Greyman, Thank you very much for reply. But unfortunately even after adding these lines in the file 'channel.c' didn't help me. It gives 'undefined reference' for all curl functions. Should I check availability of any other file for these errors? Thanks Regards Prashant Sharma On Mon, Mar 3, 2008 at 2:53 PM, Grey Man [EMAIL PROTECTED] wrote: On Mon, Mar 3, 2008 at 6:27 AM, Prashant Sharma [EMAIL PROTECTED] wrote: Hi Tilghman, Thanks for taking interest in my problem. I just want to send a http post request to my website without changing the dial plan. So I have added slightly modified http post code and some other code to channel.c got from curl/curl.h. After adding the code I compiled the asterisk code and got the error: channel.o(.text+0x): channel.c:: undefined reference to 'curl_global_init' Try adding the following above the first include in channel.c. It's what I've done in app_dial.c to get the Curl library to link properly. /*** MODULEINFO dependCurl/depend ***/ Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
Hi, I'm trying to make asterisk detect some DTMF digits during a call and post them (can't use WaitExten or Features.conf). Regards, Prashant On Mon, Mar 3, 2008 at 6:23 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Monday 03 March 2008 00:27:32 Prashant Sharma wrote: I just want to send a http post request to my website without changing the dial plan. This doesn't make any sense. The dialplan is among the easiest and least bug-prone ways of adding a curl POST call. Why would you want to code this directly into channel.c, where it does not belong, instead of making a very simple change to your dialplan? Don't explain the mechanics. Explain why you're doing it. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 registration problem
Quoting Jaap Winius [EMAIL PROTECTED]: My problem is that normal SPA3102 configurations just don't seem to work. I can't even get the FXS port to register. I'm beginning to suspect that my unit is defective. Today I called the vendor (voipsolutions.be) and was passed on to a knowledgeable tech support guy (!) who suggested that I configure a static IP address for the Internet gateway on the SPA3102 and use that instead of the LAN gateway. It worked! The registration problem is likely a bug, albeit an interesting one. Unfortunately, I'm still no better off using this device as a PSTN gateway than I am with the SPA3000, as I still can't get it to pass on the Caller ID. Cheers Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
On Monday 03 March 2008 07:18, Prashant Sharma wrote:
I'm trying to make asterisk detect some DTMF digits during a call and post
them (can't use WaitExten or Features.conf).
I would suggest that you implement that in logger.c and configure a line to
send logs to an HTTP POST (via logger.conf), with the
pbx_substitute_variables_helper function, using the ${CURL()} function
directly. You may need to preload = func_curl.so in modules.conf, but
that will work well.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
I can provide a free DID in the US if you need one. Although I would need to charge usage at $0.09 in and $0.019 out. Let me know Cp Corey Potts 480-889-7590 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Sunday, March 02, 2008 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID number On Sun, Mar 2, 2008 at 3:21 AM, Mike [EMAIL PROTECTED] wrote: Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. Hey Mike - give IPKall a try: http://www.ipkall.com/ They'll give you a free Washington state DID along with free SIP to your asterisk server. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
On Mon, Mar 03, 2008 at 11:13:28AM -0600, Tilghman Lesher wrote:
On Monday 03 March 2008 07:18, Prashant Sharma wrote:
I'm trying to make asterisk detect some DTMF digits during a call and post
them (can't use WaitExten or Features.conf).
I would suggest that you implement that in logger.c and configure a line to
send logs to an HTTP POST (via logger.conf), with the
pbx_substitute_variables_helper function, using the ${CURL()} function
directly. You may need to preload = func_curl.so in modules.conf, but
that will work well.
Or a simple log watcher.
tail -n0 -f /var/log/asterisk/debug | \
grep 'DTMF digit: [0-9#*]' | \
your_custum_filter
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
That's almost certainly your problem. When you run sidecars with the Polycom 601, you can't rely on PoE - there isn't enough power supplied. Connect your powerpack to the phone and the problem should go away. Semi random reboots are not uncommon on the 601 with sidecars if you're running it on PoE. Well, I wish it were that easy. Really, I do!!! I put a 601 power supply on Friday afternoon. Have have had 2 reboots already this morning during pages. The 601 simply can't handle the traffic of 23 simultaneous Buddy Watch updates. If a call comes in during a page. It will crash every time. We're getting a 650 in to see if that will fix the problem (as it did for others) Thanks Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem transferring calls some of the times
Ian (and the rest of the list), I've found something in order to resolve this issue... In the config file (sample) features.conf are some commented lines that said: *; Note that the DTMF features listed below only work when two channels have answered and are bridged together. ; They can not be used while the remote party is ringing or in progress. If you require this feature you can use ; chan_local in combination with Answer to accomplish it.* I will try this and let you know anything new about this issue, If you (or anyone) can try it too and if this fix the issue a post with the config is really appreciated. -- Raul Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom VSX 7000e Series Asterisk
Anyone have any experience tying the Polycom VSX 7000e Asterisk together? It says it supports standards based SIP servers but thought I'd see if anyone had real world experience. Thanks, Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem transferring calls some of the times
In the config file (sample) features.conf are some commented lines that said: *; Note that the DTMF features listed below only work when two channels have answered and are bridged together. ; They can not be used while the remote party is ringing or in progress. If you require this feature you can use ; chan_local in combination with Answer to accomplish it.* BTW: I don't have a clue how *can I use chan_local in combination with Answer to accomplish it.*, so if anyone knows please give some help! Thanks in advance... -- Raul Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
http://vitelity.net has 800# DIDs for $0.50/month plus usage (which is like $0.02/min I think)This price has been very bearable for me to just experiment with -- I can ask anyone I want to call me to test my services and they don't have to worry about toll charges Moj Mike wrote: hey Folks, Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. I am experimenting with asterisk, for home use. thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1, Rhino, Nortel
Hi all, I'm trying to insert a Rhino Ceros box equipped with a Rhino R2T1 dual-T1 card and running the latest version of Trixbox (2.4.2) between the central office and a Nortel Option 11. The switch at the CO is a DMS100. Basically, I'm taking the T1, connecting it to port 0 on the R2T1 card, and then connecting port 1 to the Nortel. (Actually a CSU and then the Nortel) We're running PRI over T1... Channels 1-23 are B and channel 24 is D. So I configured the system ahead of time with line encoding, line length, switchtype, timing source, etc. The timing source on port 0 in Zaptel.conf is '1' so I get timing from the CO and it is '0' on port 1 so I send timing to the Nortel. When I hooked it up over the weekend, the spans came up as expected with no errors or anything. Calls between the Asterisk box and the CO work like a charm. The CO doesn't know it's talking to a different box and I get everything I need, call ID, DID, etc with no problems at all. But the calls between the Asterisk box and the Nortel will not go through. I enabled debug on that span and placed calls both ways. When I call from the Nortel to the Asterisk box, the PRI debug shows the call failed with cause code 100. Based on what I can find, this looks like the Nortel is mad about the formatting of something in the messages. When I reverse that and call from the Asterisk box to the Nortel, those calls fail with a cause code of 54. Best I can tell that means 'incoming call barred' but how could it be barred? The Asterisk box should look like the DMS100 to the Nortel. I duplicated the calling information I was seeing from the CO when I tried to call the Nortel plus I tried a couple of variants... no dice. Am I missing something here? I don't understand how I can be talking to a real DMS100 on one T1 and it works perfect but when I act like a DMS100 on the other T1, the Nortel is getting mad. Can anyone offer some ideas? Maybe a clarification on these cause codes? My depth of knowledge in this area isn't that deep... a wading pool at best... so I'm hoping one of you guys that has worked with this stuff a long time might be able to give me some direction. I'm posting below Zaptel, Zapata, and a CLI dump of a call from the Nortel into the Asterisk system with pri debug span turned on. TIA! Jason # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 2: R2T1/0/1 R2T1 (PCI) Card 0 Span 1 span=2,1,0,esf,b8zs # termtype: cpe bchan=1-23 dchan=24 # Span 3: R2T1/0/2 R2T1 (PCI) Card 0 Span 2 span=3,0,0,esf,b8zs # termtype: net bchan=25-47 dchan=48 # Span 4: Rhino RCB8FXX/1 Rhino RCB8FXX/1 fxsks=49 fxsks=50 fxoks=51 fxoks=52 # ??: 53 ---/1/4 # ??: 54 ---/1/5 # ??: 55 ---/1/6 # ??: 56 ---/1/7 # Global data loadzone= us defaultzone = us ; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 2: R2T1/0/1 R2T1 (PCI) Card 0 Span 1 group=0 context=from-trunk switchtype = dms100 signalling = pri_cpe channel = 1-23 ; Span 3: R2T1/0/2 R2T1 (PCI) Card 0 Span 2 group=1 context=from-trunk switchtype = dms100 signalling = pri_net channel = 25-47 ; Span 4: Rhino RCB8FXX/1 Rhino RCB8FXX/1 ;;; line=49 FXO/1/0 signalling=fxs_ks callerid=asreceived group=3 context=from-pstn channel = 49 context=default ;;; line=50 FXO/1/1 signalling=fxs_ks callerid=asreceived group=3 context=from-pstn channel = 50 context=default ;;; line=51 FXS/1/2 signalling=fxo_ks callerid=Channel 51 6051 mailbox=6051 group=5 context=from-internal channel = 51 callerid= mailbox= group= context=default ;;; line=52 FXS/1/3 signalling=fxo_ks callerid=Channel 52 6052 mailbox=6052 group=5 context=from-internal channel = 52 callerid= mailbox= group= context=default ; ??: 53 ---/1/4 ; ??: 54 ---/1/5 ; ??: 55 ---/1/6 ; ??: 56 ---/1/7 Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 1 (reference 21/0x15) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 04 e9 80 83 15] Channel ID (len= 6) [ Ext: 1 IntID: Explicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 0
[asterisk-users] Aastra phones and park/pickup feature
We are installing Aastra phones (480's and 57i's) into a fairly simple asterisk setup. Although call park pickup work fine using xfer to 700 (to park), dial 701 (to pickup), we are unable to make the park/pickup softkey feature work on the aastra's. Although we've programmed the softkeys per the manuals, they seem to have no effect (just dead). For example, our 57i is setup like this: softkey4 type: park softkey4 label: Park softkey4 value: asterisk;70 softkey4 line: 1 softkey4 states: connected softkey4 type: pickup softkey4 label: Pickup softkey4 value: asterisk;70 softkey4 value: 1 softkey4 states: idle, outgoing (we also tried asterisk;700 with the same result). Has anyone got the softkey park/pickup working on aastra? Thanks Michelle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra phones and park/pickup feature
You'll want to use the XML park and pickup with the aastras. Feel free to ping me off list if you need help. -Darren Dwright at d2-tech dot com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Monday, March 03, 2008 2:45 PM To: 'Asterisk Users List' Subject: [asterisk-users] Aastra phones and park/pickup feature We are installing Aastra phones (480's and 57i's) into a fairly simple asterisk setup. Although call park pickup work fine using xfer to 700 (to park), dial 701 (to pickup), we are unable to make the park/pickup softkey feature work on the aastra's. Although we've programmed the softkeys per the manuals, they seem to have no effect (just dead). For example, our 57i is setup like this: softkey4 type: park softkey4 label: Park softkey4 value: asterisk;70 softkey4 line: 1 softkey4 states: connected softkey4 type: pickup softkey4 label: Pickup softkey4 value: asterisk;70 softkey4 value: 1 softkey4 states: idle, outgoing (we also tried asterisk;700 with the same result). Has anyone got the softkey park/pickup working on aastra? Thanks Michelle This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
Does this allow unlimited ports and usages only. I would love this kind of arrangement with 4 ports. Cheers Andrew Ladanowski AddInSolutions Inc. www.addinsol.com [EMAIL PROTECTED] Phone: 954-815-2402 Fax: 954-414-8432 CONFIDENTIAL : The information in this email (including any attachments) is confidential and may be privileged. If you are not the intended recipient, you may not and must not read, print, forward, use or disseminate the information contained herein. Although this email (and any attachments) are believed to be free of any virus or other defect that might affect any computer system into which it is received and opened, it is the responsibility of the recipient to ensure that it is free of viruses or defects and no responsibility is accepted by the sender for any loss or damage arising or resulting in any way from its receipt or use. If you are not the intended recipient of this message, please reply to the sender and include this message and then delete this message from your inbox and your archive and/or discarded messages files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Corey Potts Sent: Monday, March 03, 2008 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID number I can provide a free DID in the US if you need one. Although I would need to charge usage at $0.09 in and $0.019 out. Let me know Cp Corey Potts 480-889-7590 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Sunday, March 02, 2008 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID number On Sun, Mar 2, 2008 at 3:21 AM, Mike [EMAIL PROTECTED] wrote: Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. Hey Mike - give IPKall a try: http://www.ipkall.com/ They'll give you a free Washington state DID along with free SIP to your asterisk server. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Mime-construct, exim4 and job numbers
Hi, I'm using mime-construct to send fax files (from an Asterisk server). mime-construct is using exim4 to have its job done. Trouble is I'm not pleased with the way I can link mime-construct and exim4 jobs (whenever an exim4 job fails, mime-construct exit code is still 0) : all I'm doing at the moment is reading exim4 and syslog log files for events matching. In syslog, I've got : 2008-03-03 10:53:02 hostname emailfax: sending file foo.tiff to [EMAIL PROTECTED] produced 0 exit code In exim4, I've got : 2008-03-03 10:53:02 1JWHdV-0006g8-1u = [EMAIL PROTECTED] U=root P=local S=24891 So I guess, this foo.tiff file were sent with this 1JWHdV-0006g8-1u exim4 message. I'm looking for a way to have in syslog : 2008-03-03 10:53:02 hostname emailfax: sending file foo.tiff to [EMAIL PROTECTED] produced 0 exit code using 1JWHdV-0006g8-1u exim4 message How can I teach : - exim4 to report mime-construct this 1JWHdV-0006g8-1u code - and mime-construct to log it My understanding of Perl (mime-construct is written in Perl) is poor. Regards PS: I warned that this message is really Off-Topic but I know a lot of talented Linux administrators read this mailing list, so ... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Switchvox feedback
I have a customer that wants to get switchvox, since I have never used it, I would like to hear some feedback from active users of switchvox. In specific: 1. Does it use realtime or conf files 2. Is it possible to change it manually? 3. Is SSH access to login to console/shell available? 4. Are you or your customers happy with the user interface? TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ekiga sip registration fails; externip no help
ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=as64618445 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 113 REGISTER User-Agent: Asterisk PBX 1.6.0-beta4 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- --- SIP read from UDP://86.64.162.35:5060 --- SIP/2.0 406 Not Acceptable Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport=5060;received=96.xxx.253.yy From: sip:[EMAIL PROTECTED];tag=as64618445 To: sip:[EMAIL PROTECTED];tag=12d18c5009a2de32fca8ae9d70ad0321.ef55 Call-ID: [EMAIL PROTECTED] CSeq: 113 REGISTER Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 86.64.162.35:5060 Noisy feedback tells: pid=24578 req_src_ip=96.xxx.253.yy req_src_port=5060 in_uri=sip:ekiga.net out_uri=sip:ekiga.net via_cnt==1 - --- (9 headers 0 lines) --- -- Got SIP response 406 Not Acceptable back from 86.64.162.35 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switchvox feedback
On Mon, Mar 3, 2008 at 4:35 PM, C F [EMAIL PROTECTED] wrote: I have a customer that wants to get switchvox, since I have never used it, I would like to hear some feedback from active users of switchvox. In specific: 1. Does it use realtime or conf files 2. Is it possible to change it manually? 3. Is SSH access to login to console/shell available? 4. Are you or your customers happy with the user interface? TIA First of all SwitchVox is a GREAT product. Second, I would boot to G4U and make a bit level clone of your system and dump that off on the network somewhere before messing around. I beleive (and I could be wrong) that it uses conf files. Anything is possible. At bootup select a and add single to get to # prompt. Either add a new user or change root's password. I do no think that root is enabled to login via SSH. I just installed webmin (I love webmin) on mine and took back the control that was locked down. To that end, I don't see why you would be locked to Digium hardware either since you could log in as root and install other manuafacturer's patches or Drivers. Updating might be painful down the road though. Also, peek around for any phone home scripts or OpenVPN, I am not saying there are any, I am just curious since I do not have access to any SwitchVox boxen anymore. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1, Rhino, Nortel
On Mon, Mar 3, 2008 at 2:19 PM, Gleim, Jason [EMAIL PROTECTED] wrote: Hi all, I'm trying to insert a Rhino Ceros box equipped with a Rhino R2T1 dual-T1 card and running the latest version of Trixbox (2.4.2) between the central office and a Nortel Option 11. The switch at the CO is a DMS100. Basically, I'm taking the T1, connecting it to port 0 on the R2T1 card, and then connecting port 1 to the Nortel. (Actually a CSU and then the Nortel) We're running PRI over T1... Channels 1-23 are B and channel 24 is D. So I configured the system ahead of time with line encoding, line length, switchtype, timing source, etc. The timing source on port 0 in Zaptel.conf is '1' so I get timing from the CO and it is '0' on port 1 so I send timing to the Nortel. When I hooked it up over the weekend, the spans came up as expected with no errors or anything. Calls between the Asterisk box and the CO work like a charm. The CO doesn't know it's talking to a different box and I get everything I need, call ID, DID, etc with no problems at all. But the calls between the Asterisk box and the Nortel will not go through. I enabled debug on that span and placed calls both ways. When I call from the Nortel to the Asterisk box, the PRI debug shows the call failed with cause code 100. Based on what I can find, this looks like the Nortel is mad about the formatting of something in the messages. When I reverse that and call from the Asterisk box to the Nortel, those calls fail with a cause code of 54. Best I can tell that means 'incoming call barred' but how could it be barred? The Asterisk box should look like the DMS100 to the Nortel. I duplicated the calling information I was seeing from the CO when I tried to call the Nortel plus I tried a couple of variants... no dice. Am I missing something here? I don't understand how I can be talking to a real DMS100 on one T1 and it works perfect but when I act like a DMS100 on the other T1, the Nortel is getting mad. Can anyone offer some ideas? Maybe a clarification on these cause codes? My depth of knowledge in this area isn't that deep... a wading pool at best... so I'm hoping one of you guys that has worked with this stuff a long time might be able to give me some direction. I'm posting below Zaptel, Zapata, and a CLI dump of a call from the Nortel into the Asterisk system with pri debug span turned on. TIA! Jason # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 2: R2T1/0/1 R2T1 (PCI) Card 0 Span 1 span=2,1,0,esf,b8zs # termtype: cpe bchan=1-23 dchan=24 # Span 3: R2T1/0/2 R2T1 (PCI) Card 0 Span 2 span=3,0,0,esf,b8zs # termtype: net bchan=25-47 dchan=48 # Span 4: Rhino RCB8FXX/1 Rhino RCB8FXX/1 fxsks=49 fxsks=50 fxoks=51 fxoks=52 # ??: 53 ---/1/4 # ??: 54 ---/1/5 # ??: 55 ---/1/6 # ??: 56 ---/1/7 # Global data loadzone= us defaultzone = us ; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 2: R2T1/0/1 R2T1 (PCI) Card 0 Span 1 group=0 context=from-trunk switchtype = dms100 signalling = pri_cpe channel = 1-23 ; Span 3: R2T1/0/2 R2T1 (PCI) Card 0 Span 2 group=1 context=from-trunk switchtype = dms100 signalling = pri_net channel = 25-47 ; Span 4: Rhino RCB8FXX/1 Rhino RCB8FXX/1 ;;; line=49 FXO/1/0 signalling=fxs_ks callerid=asreceived group=3 context=from-pstn channel = 49 context=default ;;; line=50 FXO/1/1 signalling=fxs_ks callerid=asreceived group=3 context=from-pstn channel = 50 context=default ;;; line=51 FXS/1/2 signalling=fxo_ks callerid=Channel 51 6051 mailbox=6051 group=5 context=from-internal channel = 51 callerid= mailbox= group= context=default ;;; line=52 FXS/1/3 signalling=fxo_ks callerid=Channel 52 6052 mailbox=6052 group=5 context=from-internal channel = 52 callerid= mailbox= group= context=default ; ??: 53 ---/1/4 ; ??: 54 ---/1/5 ; ??: 55 ---/1/6 ; ??: 56 ---/1/7 Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 1 (reference 21/0x15) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)
Re: [asterisk-users] DID number
Please send info regarding free incoming DID's Andrew Ladanowski AddInSolutions Inc. www.addinsol.com [EMAIL PROTECTED] Phone: 954-815-2402 Fax:954-414-8432 CONFIDENTIAL: The information in this email (including any attachments) is confidential and may be privileged. If you are not the intended recipient, you may not and must not read, print, forward, use or disseminate the information contained herein. Although this email (and any attachments) are believed to be free of any virus or other defect that might affect any computer system into which it is received and opened, it is the responsibility of the recipient to ensure that it is free of viruses or defects and no responsibility is accepted by the sender for any loss or damage arising or resulting in any way from its receipt or use. If you are not the intended recipient of this message, please reply to the sender and include this message and then delete this message from your inbox and your archive and/or discarded messages files. Thank you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Corey Potts Sent: Monday, March 03, 2008 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID number I can provide a free DID in the US if you need one. Although I would need to charge usage at $0.09 in and $0.019 out. Let me know Cp Corey Potts 480-889-7590 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Sunday, March 02, 2008 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID number On Sun, Mar 2, 2008 at 3:21 AM, Mike [EMAIL PROTECTED] wrote: Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. Hey Mike - give IPKall a try: http://www.ipkall.com/ They'll give you a free Washington state DID along with free SIP to your asterisk server. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP600 + PC share same switch port withVLAN
James, thanks for the suggestion. I am just using native vlan and I did what you said and I believe it works :-) interface FastEthernet2/0/2 description VOIP VLAN 100 switchport trunk encapsulation dot1q switchport trunk allowed vlan 1,100 switchport mode trunk duplex full speed 100 However, when I entered spanning-tree portfast trunk, I received a warning message which says: Warning: portfast should only be enabled on ports connected to a single host. Connecting hubs, concentrators, switches, bridges, etc... to this interface when portfast is enabled, can cause temporary bridging loops. Use with CAUTION As the polycom phone is acting like a switch, I decided not to put that option in. ***Also, can I just confirm that with the current QOS (quality of service) settings on the polycom phones, the phone should have priority over the PC? Thanks in advance. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Sneeringer Sent: Saturday, 1 March 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP600 + PC share same switch port withVLAN As far as I can tell, with Polycom phones you cannot do what you're asking (which is for the PC and the phone to be in the same VLAN while the PC is connected to the phone). I don't know how they handle it when the voice frames are untagged, but they definitely won't pass tagged voice frames to the PC port: http://knowledgebase.polycom.com/KanisaPlatform/Publishing/616/12526_f.S AL _PUBLIC_1_2.html Your switch port is configured for untagged frames on a single VLAN (that's what access mode is). Polycom phones need voice and data to be on separate VLANs in order for you to use the PC port. Since Polycom phones apparently don't support the Cisco Voice VLAN feature, you need to configure the port as a trunk port, which will allow you to send multiple VLANs to the phone. The phone will take frames tagged for your designated voice VLAN, and will pass the rest on to the PC port. For example: interface FastEthernet2/0/1 switchport trunk encapsulation dot1q switchport trunk native vlan XXX switchport trunk allowed vlan XXX,100 switchport mode trunk spanning-tree portfast trunk Replace XXX with whatever your PC VLAN is. Setting XXX as the native VLAN for this port will cause frames in that VLAN to be untagged for that port, which is what your PC probably expects. If it happens to be 1, then it's the native VLAN by default. The last command may or may not be available, depending on your version of IOS. If it isn't, portfast just won't work and you're just stuck with STP negotiation anytime the port bounces. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
On Tue, Feb 26, 2008 at 10:51 PM, Joshua Kinard [EMAIL PROTECTED] wrote: Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very likely, 380's as well). I just learned this the hard way. --J ...can you expand on that please ? I'm on my way to getting one of the newer Digium TE220B PCIe dual T1/E1 to put on such a system. How can I tell if the card has a TigerJet chipset ? -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
I've used lots of Digium T1 cards on DL380 / DL320's without a hiccup. -Darren From: [EMAIL PROTECTED] on behalf of Joshua Kinard Sent: Tue 2/26/2008 5:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very likely, 380's as well). I just learned this the hard way. --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Norman Franke Sent: Tuesday, February 26, 2008 5:27 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Had it with Dell Garbage On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED] wrote: On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote: I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? HP DL380 is my baby. Thanks, Steve Totaro Ditto. We've been using HPs for a while without problem. I'm currently using a DL380 (a recent quad processor one) and it screams. -Norman This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
On Monday 03 March 2008 21:05:43 Ex Vito wrote: On Tue, Feb 26, 2008 at 10:51 PM, Joshua Kinard [EMAIL PROTECTED] wrote: Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very likely, 380's as well). I just learned this the hard way. ...can you expand on that please ? I'm on my way to getting one of the newer Digium TE220B PCIe dual T1/E1 to put on such a system. How can I tell if the card has a TigerJet chipset ? None of the current cards are based on TigerJet chipsets. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
Hi,
Thanks but using the logger.c approach will allow the IVR to receive the
digits in case 's' extension answers the call. That might result in the dial
plan dialing an extension or going to the 'i' extension and hanging up.
Ssorry about the confusion.
Thanks Regards
Prashant Sharma
On Mon, Mar 3, 2008 at 10:43 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 03 March 2008 07:18, Prashant Sharma wrote:
I'm trying to make asterisk detect some DTMF digits during a call and
post
them (can't use WaitExten or Features.conf).
I would suggest that you implement that in logger.c and configure a line
to
send logs to an HTTP POST (via logger.conf), with the
pbx_substitute_variables_helper function, using the ${CURL()} function
directly. You may need to preload = func_curl.so in modules.conf, but
that will work well.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem transferring calls some of the times
Hi Raul I have bypassed my Grandstream's transfer function, by enabling *2 transfers in features.conf, and setting canreinvite=no in sip.conf Hope this helps you Ian Raúl Gómez C. said the following on 03-Mar-08 08:34 PM: In the config file (sample) features.conf are some commented lines that said: /; Note that the DTMF features listed below *only work when two channels have answered and are bridged together*. ; They *can not be used while the remote party is ringing or in progress*. If you require this feature you can use ; chan_local in combination with Answer to accomplish it./ BTW: I don't have a clue how /can I use chan_local in combination with Answer to accomplish it./, so if anyone knows please give some help! Thanks in advance... -- Raul Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
