Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected
Does, it not look like a privilege problem on the file /usr/local/bin/wmix? Please check that also. Rgds SIP - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 31, 2003 10:30 PM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Yes. It works even in the macro (not only oin the shell), but because of that warning, the macro exit. The resulting file of the mix is perfect (tried after that), but still that warning. See my previous mail ... testet with a simple 'ls' command. I don't know what to do...:( BR, Dan - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Thursday, July 31, 2003 7:49 PM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Try to do the same in shell. Does it work ? Martin On Thu, 31 Jul 2003, Dan wrote: Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple wmix without any parameter and I get the same error. . -- Executing System(SIP/103-b7c0, /usr/local/bin/wmix /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav /var/spool/asterisk/monitor/31072003-19:08:11-103.wav) in new stack WARNING[1200825920]: File app_system.c, Line 57 (skel_exec): Unable to execute '/usr/local/bin/wmix /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav /var/spool/asterisk/monitor/31072003-19:08:11-103.wav' == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on 'SIP/103-b7c0' in macro 'record-cleanup' == Spawn extension (fullaccess, h, 1) exited non-zero on 'SIP/103-b7c0' Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected
Hi, It is the same with any other application I try to run from the System() application. I don't think is a privilege problem. Even with 'ls' as command, it displays the directory listing and then exit with error. Thanks, Dan - Original Message - From: Sip Rtp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 10:07 AM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Does, it not look like a privilege problem on the file /usr/local/bin/wmix? Please check that also. Rgds SIP - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 31, 2003 10:30 PM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Yes. It works even in the macro (not only oin the shell), but because of that warning, the macro exit. The resulting file of the mix is perfect (tried after that), but still that warning. See my previous mail ... testet with a simple 'ls' command. I don't know what to do...:( BR, Dan - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Thursday, July 31, 2003 7:49 PM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Try to do the same in shell. Does it work ? Martin On Thu, 31 Jul 2003, Dan wrote: Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple wmix without any parameter and I get the same error. . -- Executing System(SIP/103-b7c0, /usr/local/bin/wmix /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav /var/spool/asterisk/monitor/31072003-19:08:11-103.wav) in new stack WARNING[1200825920]: File app_system.c, Line 57 (skel_exec): Unable to execute '/usr/local/bin/wmix /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav /var/spool/asterisk/monitor/31072003-19:08:11-103.wav' == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on 'SIP/103-b7c0' in macro 'record-cleanup' == Spawn extension (fullaccess, h, 1) exited non-zero on 'SIP/103-b7c0' Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 29 July 2003 13:47, Tais M. Hansen wrote: I havn't used the h323 channel of Asterisk for a while, but today I needed to test a few things only I found out that Asterisk/H323 crashes my Siemens optipoint 400 phone. It seems to be the audio codecs that's causing it. Is something broken in chan_h323? To follow up on this question, I flashed the Siemens ip phone with a really really old image and forced G711 alaw. This made it behave nicely. - -- Regards, Tais M. Hansen ComX -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/KhWM2TEAILET3McRArR9AJ4hdDwaUtvEuLpKAwHH5m1joTfF2QCcDIM1 LaErjBZO5z1VD0Rti9L8khU= =C18H -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 over NAT
hello I am trying to configure h323 over NAT Can some one help me out Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected
On Fri, 2003-08-01 at 09:14, Dan wrote:
Hi,
It is the same with any other application I try to run from the System()
application.
I don't think is a privilege problem. Even with 'ls' as command, it displays
the directory listing and then exit with error.
If you create mysystem.c containing:
#include stdlib.h
#include stdio.h
int main() {
int ret;
ret = system(/bin/ls /dev/null);
printf(system(\/bin/ls /dev/null\) returned %d\n, ret);
return(ret);
}
and compile it with:
gcc mysystem.c -o mysystem
and run:
./mysystem
what is the output?
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
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Re: [Asterisk-Users] h323 over NAT
H.323 doesn't deal with NAT. Jeremy McNamara ayaz wrote: hello I am trying to configure h323 over NAT Can some one help me out Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP with an iptables fiewall
Am I the only person in the * world who can't get a sip connection through an iptables firewall? I've got everything else working fine. Xten - PSTN, Xten - Analog, IAX - IAX, but exten = 3733,1,Dial(SIP/[EMAIL PROTECTED]) ; evades me, ngrep @ port 5060 says the INVITES go out but how do I get something back? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected
Hi,
This is the result:
[EMAIL PROTECTED] temp]# ./mysystem
system(/bin/ls /dev/null) returned 0
[EMAIL PROTECTED] temp]#
BR,
Dan
- Original Message -
From: Armand A. Verstappen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 01, 2003 10:41 AM
Subject: Re: [Asterisk-Users] 'System' application exit with error even ifit
performs the job as expected
If you create mysystem.c containing:
#include stdlib.h
#include stdio.h
int main() {
int ret;
ret = system(/bin/ls /dev/null);
printf(system(\/bin/ls /dev/null\) returned %d\n, ret);
return(ret);
}
and compile it with:
gcc mysystem.c -o mysystem
and run:
./mysystem
what is the output?
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
___
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[Asterisk-Users] Delete voicemail files on disk
Hi everyone, I know there isn't an option in the config to only mail voicemail files and not keep them stored on disk, but is it save to run a nightly cron job at 3 A.M. just wiping the files that are in the INBOX subfolder ? Greetings, Tj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP calls cause Segmentation Fault
Mark, the server has already been installed at a client and the only access to internet I have is from behind a NAT therefore I cannot give you access to log into the server. Also, I do not have an IRC client on the machine, and the closest windows machine is 4 floors away. What is the procedure to extract debug data I can send you please ? thanks Dave - Original Message - From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 31, 2003 8:11 PM Subject: Re: [Asterisk-Users] SIP calls cause Segmentation Fault Yes, find me on #asterisk so I can login. Be sure you're generating cores and running on very latest CVS. Mark On Thu, 31 Jul 2003, Dave Alan Caruana wrote: I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan is a modification of the sample one that came with Asterisk, and I haven't touched that many other conf files actually. Any way I can get this debugged? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] memory leak?
hi all seems there's a memory leak in asterisk somewhere. probably in chan_mgcp. It seems like 2048kB is allocated but not released each time I lift the handset. This is, however, never released. So... oslpbx:~/debug/20030801112939# ps axfv|grep asterisk | tail -1 3206 pts/0S 0:00 0 410 35601 6024 1.1 \_ /usr/sbin/asterisk -gcp (lift handset) oslpbx:~/debug/20030801112939# ps axfv|grep asterisk | tail -1 3212 pts/0S 0:00 2 410 37669 6072 1.1 \_ /usr/sbin/asterisk -gcp (hangup) oslpbx:~/debug/20030801112939# ps axfv|grep asterisk | tail -1 3206 pts/0S 0:00 0 410 37669 6076 1.1 \_ /usr/sbin/asterisk -gcp I don't know the code in asterisk that well... I tried stracing it, but I couldn't find anything roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 2254 5070 (work) +47 9801 3356 (mobile) Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP calls cause Segmentation Fault
Thanks Adam, I will try it out. cheers Dave - Original Message - From: Adam Donnison [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 1:18 AM Subject: Re: [Asterisk-Users] SIP calls cause Segmentation Fault I actually found this same thing, and traced it down to app_dial.c line 190. It doesn't explicitly check for a valid chan before trying to use it and it segfaults when it does a strlen on a chan entity. I simply put a check in that winner was non-zero before comparing it to o-chan: if (winner winner == o-chan) Adam Dave Alan Caruana wrote: I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan is a modification of the sample one that came with Asterisk, and I haven't touched that many other conf files actually. Any way I can get this debugged? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Adam Donnison email: [EMAIL PROTECTED] Saki Computer Services Pty. Ltd. 93 Kallista-Emerald Roadphone: +61 3 9752 1512 THE PATCH VIC 3792AUSTRALIAfax: +61 3 9752 1098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF modes and external IVR systems over ISDN
Hello, I'm trying to understand why when I make a call from a SIP phone to an external number who has an IVR system in which I've to choose some options using the dialpad, it does'nt recognise the key pressed and remains still waiting for my choose. I'm tryng using Grandstream 102, and i've tryed with all the 3 modes possibile: Dtmf inband, rfc2833 and INFO (obviously configuring'em also in sip.conf). I think that's a problem of the ISDN line (used with I4L modem drivers), that does'nt pass the DTMF to the end point of the call. Is there some thing i can do on the conf files or I have to search the answer into the source tree? I googlized a bit and found some patch to isdn_tty (that should solve the problem of receiving dtmf from isdn line when calling with a person), but unfortunately, the kernel version is different and also correcting a bit the patch, it doesn't work for me - i still get errors with patch. Nothing found for the contrary use: sending dtmf over the isdn to make possibile a choice into an external ivr or menu system. Any link/patch/suggestion? Without it, my * box will be soon cut off by my boss (i need to set-up our pbx ASAP - prior to holidays... :( ) -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 over NAT
Jeremy McNamara wrote: H.323 doesn't deal with NAT. Jeremy McNamara ayaz wrote: hello I am trying to configure h323 over NAT Can some one help me out Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users As far as I know there is one posibility if the equipment who is dealing with NAT can do so called H323 NAT... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP bug with Net2Phone
When I try call to net2pohe sip service in my debug I look next: We're at 192.0.0.0 port 27916 Answering with preferred capability 1 Answering with preferred capability 2 Answering with preferred capability 256 Answering with capability 4 Answering with capability 8 Answering with capability 16 Answering with capability 32 Answering with capability 64 Answering with capability 128 Answering with capability 512 Answering with capability 1024 Answering with capability 2048 Answering with capability 4096 Answering with capability 8192 Answering with capability 16384 Answering with capability 32768 10 headers, 17 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2 From: [EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as26712c28 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 384 v=0 o=root 21604 21604 IN IP4 192.0.0.0 s=session c=IN IP4 192.0.0.0 t=0 0 m=audio 27916 RTP/AVP 4 3 18 0 8 14 5 10 7 110 97 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:14 MPA/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 (no NAT) to 66.33.146.12:5060 -- Called [EMAIL PROTECTED] Sip read: SIP/2.0 407 Unauthorized Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2 From: [EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as26712c28 To: sip:[EMAIL PROTECTED];tag=3f2a5b8b-12006 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: net2phone sip:66.33.146.12:5060 User-Agent: Asterisk PBX Proxy-Authenticate: Digest realm=net2phone,nonce=55895A5A2566A49758E30C701D17BD49 Content-Length: 0 10 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2 From: [EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as26712c28 To: sip:[EMAIL PROTECTED];tag=3f2a5b8b-12006 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 66.33.146.12:5060 We're at 192.0.0.0 port 27916 Answering with preferred capability 1 Answering with preferred capability 2 Answering with preferred capability 256 Answering with capability 4 Answering with capability 8 Answering with capability 16 Answering with capability 32 Answering with capability 64 Answering with capability 128 Answering with capability 512 Answering with capability 1024 Answering with capability 2048 Answering with capability 4096 Answering with capability 8192 Answering with capability 16384 Answering with capability 32768 Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2 From: [EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as26712c28 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username=, realm=net2phone, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=55895A5A2566A49758E30C701D17BD49, response=bd4841816ed727ed12c3bc4d1b19e7a5 Content-Type: application/sdp Content-Length: 384 v=0 o=root 21586 21586 IN IP4 192.0.0.0 s=session c=IN IP4 192.0.0.0 t=0 0 m=audio 27916 RTP/AVP 4 3 18 0 8 14 5 10 7 110 97 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:14 MPA/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 (no NAT) to 66.33.146.12:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2 From: [EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as26712c28 To: sip:[EMAIL PROTECTED];tag=3f2a5b8c-12006 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Contact: net2phone sip:66.33.146.12:5060 User-Agent: Asterisk PBX Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2 From: [EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as26712c28 To: sip:[EMAIL PROTECTED];tag=3f2a5b8c-12006 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Contact: net2phone sip:66.33.146.12:5060 User-Agent: Asterisk PBX Content-Length: 240 Content-Type: application/sdp v=0 o=Net2Phone 562767273 562767273 IN IP4 66.33.136.130 s=Net2Phone c=IN IP4 66.33.136.130 t=0 0 m=audio 20182 RTP/AVP 4 101 a=ptime:90 a=rtpmap:4 G723/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 10 headers, 11 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2 From: [EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as26712c28 To: sip:[EMAIL PROTECTED];tag=3f2a5b8c-12006 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Contact: net2phone sip:66.33.146.12:5060 User-Agent: Asterisk PBX Content-Length: 240 Content-Type:
Re: [Asterisk-Users] SIP with an iptables fiewall
The SIP protocol is designed in a way that makes it tough to work with NAT. The two SIP endpoints dynamically determine the ports to use for the RTP (voice) data. Port 5060 is only used for control messages. People have gotten SIP to work via a firewall (or iptables) but it's not a trivial thing. I avoid this problem by putting an Asterisk server at each location that has SIP devices and to inter-location communication via IAX (which does NOT have problems with NAT). Another way to deal with this is to run a VPN or IP tunnel between the network the SIP device is on and the network the Asterisk server is on. However, you can get very poor quality calls with this (since many VPN systems use TCP rather than UDP). On Fri, 2003-08-01 at 03:03, Dave Cotton wrote: Am I the only person in the * world who can't get a sip connection through an iptables firewall? I've got everything else working fine. Xten - PSTN, Xten - Analog, IAX - IAX, but exten = 3733,1,Dial(SIP/[EMAIL PROTECTED]) ; evades me, ngrep @ port 5060 says the INVITES go out but how do I get something back? -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension handling.
I've designed a voice menu, someone calls a certain extension, and I send them to another context via a goto, and play a background message. After playing this message can I provide them dialtone from asterisk again, in order to dial out? Regards MIKE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension handling.
Check application DISA: DISA (Direct Inward System Access) bkw On Fri, 1 Aug 2003, Michael Baird wrote: I've designed a voice menu, someone calls a certain extension, and I send them to another context via a goto, and play a background message. After playing this message can I provide them dialtone from asterisk again, in order to dial out? Regards MIKE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] segmentation fault with asterisk and OH323
Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. Someone know where the problem ? Regards Rattana
Re: [Asterisk-Users] segmentation fault with asterisk and OH323
Rattana BIV wrote: Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. More info (config files, screen log, backtrace of core file) is needed. Someone know where the problem ? Regards Rattana Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
I would suggest you try and get hold of an AVM fritz (I got one for 4 off Ebay) card and then use chan_capi.. Later.. Hello, I'm trying to understand why when I make a call from a SIP phone to an external number who has an IVR system in which I've to choose some options using the dialpad, it does'nt recognise the key pressed and remains still waiting for my choose. I'm tryng using Grandstream 102, and i've tryed with all the 3 modes possibile: Dtmf inband, rfc2833 and INFO (obviously configuring'em also in sip.conf). I think that's a problem of the ISDN line (used with I4L modem drivers), that does'nt pass the DTMF to the end point of the call. Is there some thing i can do on the conf files or I have to search the answer into the source tree? I googlized a bit and found some patch to isdn_tty (that should solve the problem of receiving dtmf from isdn line when calling with a person), but unfortunately, the kernel version is different and also correcting a bit the patch, it doesn't work for me - i still get errors with patch. Nothing found for the contrary use: sending dtmf over the isdn to make possibile a choice into an external ivr or menu system. Any link/patch/suggestion? Without it, my * box will be soon cut off by my boss (i need to set-up our pbx ASAP - prior to holidays... :( ) -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] segmentation fault with asterisk and OH323
By my phone I call H323 client (Netmeeting) I can talk. Everything is OK. But when I hangup I have the segmentation fault. These are the last asterisk log before the segmentation fault: -- Called 192.168.1.200 -- H323:193 answered CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2145 (ast_channel_bridge): Didn't get a frame from channel: CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2213 (ast_channel_bridge): Bridge stops bridging channels CAPI[contr1/26]/1 and H323:193 Something strange is, when I call the phone from netmeeting it's work perfectly. What do you think ? (you can see my oh323.conf in attach file) Rattana - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 3:07 PM Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 Rattana BIV wrote: Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. More info (config files, screen log, backtrace of core file) is needed. Someone know where the problem ? Regards Rattana Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users oh323.conf Description: Binary data
Re: [Asterisk-Users] Congestion
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 31 July 2003 16:52, Tais M. Hansen wrote:
I place a call via one ISDN carrier (or from my cellphone) through another
ISDN carrier via chan_zap into Asterisk, which then tries to dial the H323
phone. But the calling phone keeps ringing after congestion has been
called. Calling Busy() instead of Congestion() displays the same behaviour.
It seems to work though when I do something like the following:
exten = s,1,Dial(H323/ip$${ARG1}|15)
exten = s,2,Answer()
exten = s,3,Congestion()
This breaks CDRs as they get marked as ANSWERED. :( ... Why does neither h323
- - chan_h323 - chan_zap - pstn nor pstn - chan_zap - chan_h323 - h323
react to congestion? The ip phone doesn't even make a ringing tone while
calling.
- --
Regards,
Tais M. Hansen
ComX
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Re: [Asterisk-Users] SCO/Linux concerns
On Wednesday 30 July 2003 19:07, Ajit M Kallingal wrote: I am getting a bit concerned about the SCO vs IBM issue You may wish to read a recent position paper by Eben Moglen. http://osdl.org/osdlpress/2003_07_31_beaverton.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk SIP bug with Net2Phone
This may help: I had to edit chan_sip.c and change User-Agent: Asterisk PBX to Cisco ATA 186 to get n2p sip to work -Original Message- From: Kostyantyn Ahafontsev [mailto:[EMAIL PROTECTED] Sent: 01 August 2003 13:38 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk SIP bug with Net2Phone When I try call to net2pohe sip service in my debug I look next: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
WipeOut wrote: I would suggest you try and get hold of an AVM fritz (I got one for 4 off Ebay) card and then use chan_capi.. Yeah i know but unfortunately i *must* use that couple of modems I've here :-( Anyway, with a combo of firmware+conf i've been able to send dtmf tones over ISDN now. The problem that still remains is: 1) i receive a lot of dtmf tones from the line (i4l) when i speak to phone 2) some IVR system is not recognized from *, it still wait for an answer (ringing the line), even when the remote IVR has answered the call. -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c) or unixODBC (cdr_unixodbc.c) ?
I would like some Asterisk community input: It was requested of me to create an Asterisk FreeTDS (cdr_tds.c) module to allow Asterisk to populate MS SQL Server (and Sybase) with CDR records. unixODBC has been thrown around as a bright shiny button but it adds an additional layer of DB abstraction. That additional layer of abstraction can cause problems when it is not necessary for our TDS (MS SQL) usage. I generally exchange data through as few hands as possible to eliminate potential areas for problems. For cdr_tds.c Asterisk - cdr_tds.c - FreeTDS - TDS Database For cdr_unixodbc.c Asterisk - cdr_unixodbc.c - unixODBC - FreeTDS - TDS Database or Asterisk - cdr_unixodbc.c - unixODBC - (unixODBC driver for DB) - Database (Oracle, Sybase, MS SQL Server, MySQL, Postgres, etc...) FreeTDS already has a unixODBC driver (libtdsodbc.so) so for our needs we can use FreeTDS directly or through a additional abstraction with unixODBC. Currently in Asterisk is the support for a CDR CSV (cdr_csv.c), cdr_mysql.c, and I saw on the list a few weeks ago cdr_sybase.c What would help the Asterisk community the most? Is there really a demand out there for anything other than MySQL CSV flat file TDS (Sybase and MS SQL Server) I have never heard the words Oracle or Postgres thrown around so I am wondering if there is a justifiable need to support such systems. I will do TDS. That is a given. However, before I go off and create something, I want the Asterisk community input as to what their needs are. Our needs: Asterisk to populate a CDR table in MS SQL Server Community needs: ?? I agree unixODBC is a nice shiny button but the TDS need is to allow an integration with an end MS SQL billing system. Our need can still be accomplished with a unixODBC but it adds an additional layer that, quite frankly, can cause problems when we do not need it. Bright shiny buttons have always been a support problem when they are not always necessary. In addition to the programming is a readme. The readme is by far more complicated with a unixODBC setup versus configuring 2 FreeTDS config files. Opinions? Desires? Feature request? Regardless this work will be released open source to the Asterisk community. Erik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF onto the reall world
Hello All, I was originally having problems just getting DTM from the SIP phone over to asterisk for the voice mail app. But through settings in the SIP client got that working. I am know at the stage know though that asterisk doesn't appear to be passing DTMF over to the real world, I am using ISDN 4 Linux with an ISDN2e. What settings in Asterisk are responsible for this? Thanks Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP session traversing Asterisk server ...
Dear all, Concerning the REINVITE discussion... Does anybody can confirm if the Cisco ATA186 accepts the REINVITE Sip Message, and if is compatible with this feature? In other words, is it possible to make the ATA186 change the RTP destination and start sending the media packets straight to destination point instead of keep sending to Asterisk? Thanks Regards, Artur C. Severo Eng., M.Sc. Network Engineer Tel: 55 51 3328 0636 #242 Low, Adam wrote: Thanks all, I spent some time on this last night with packet sniffer in hand, the 'canreinvite' option makes sense and seems to work well for me (running latest * CVS release) when used between 79xx phones and the AS5300 gateway although I get some somewhat expected problems with 79xx that are NAT'd behind ADSL/cable connections. I don't seem to be hitting the bug that Dave mentioned below ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'System' application exit with error evenifit performs the job as expected
- Original Message -
From: Armand A. Verstappen [EMAIL PROTECTED]
If you create mysystem.c containing:
#include stdlib.h
#include stdio.h
int main() {
int ret;
ret = system(/bin/ls /dev/null);
printf(system(\/bin/ls /dev/null\) returned %d\n, ret);
return(ret);
}
gcc mysystem.c -o mysystem
./mysystem
what is the output?
On Fri, 2003-08-01 at 10:40, Dan wrote:
This is the result:
[EMAIL PROTECTED] temp]# ./mysystem
system(/bin/ls /dev/null) returned 0
[EMAIL PROTECTED] temp]#
Okay, that at least rules out the suggestion earlier in this thread that
your systems' system call is broken.
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
signature.asc
Description: This is a digitally signed message part
Re: [Asterisk-Users] 'System' application exit with error evenifit performs the job as expected
Then?
What can it be?
Thanks,
Dan
- Original Message -
From: Armand A. Verstappen [EMAIL PROTECTED]
If you create mysystem.c containing:
#include stdlib.h
#include stdio.h
int main() {
int ret;
ret = system(/bin/ls /dev/null);
printf(system(\/bin/ls /dev/null\) returned %d\n, ret);
return(ret);
}
gcc mysystem.c -o mysystem
./mysystem
what is the output?
On Fri, 2003-08-01 at 10:40, Dan wrote:
This is the result:
[EMAIL PROTECTED] temp]# ./mysystem
system(/bin/ls /dev/null) returned 0
[EMAIL PROTECTED] temp]#
Okay, that at least rules out the suggestion earlier in this thread that
your systems' system call is broken.
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
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[Asterisk-Users] Problem with SIP Native Bridging and UPnP
My configuration is comprised of two Snom 200 phones, two FXO cards connected to two PSTN lines, and one SIP account at iConnect. Snom1 has a VPN connection to the remote Asterisk server. Snom2 is using UPnP behind a Linksys WRT54G router/firewall to connect to the same server. All outgoing calls are routed to iConnect. Snom1 works correctly. Snom2 can call Snom1, and receive calls from the PSTN, but calls made to the PSTN through iConnect complete but nothing can be heard. The debug output shows that Asterisk is attempting a native bridge between iConnect and Snom2 at the time the call is answered and the line goes dead. It would seem that a native bridge is not working between iConnect and the phone behind a UPnP firewall. Is the UPnP device the problem? Can it be configured to fix the problem? If not, can native bridging be disabled for this phone? Otherwise, it would seem that I will have to establish a VPN to this phone just to solve this problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c)or unixODBC (cdr_unixodbc.c) ?
On Fri, 2003-08-01 at 09:17, Erik Anderson wrote: I would like some Asterisk community input: It was requested of me to create an Asterisk FreeTDS (cdr_tds.c) module to allow Asterisk to populate MS SQL Server (and Sybase) with CDR records. For cdr_tds.c Asterisk - cdr_tds.c - FreeTDS - TDS Database For cdr_unixodbc.c Asterisk - cdr_unixodbc.c - unixODBC - FreeTDS - TDS Database or Asterisk - cdr_unixodbc.c - unixODBC - (unixODBC driver for DB) - Database (Oracle, Sybase, MS SQL Server, MySQL, Postgres, etc...) FreeTDS already has a unixODBC driver (libtdsodbc.so) so for our needs we can use FreeTDS directly or through a additional abstraction with unixODBC. Currently in Asterisk is the support for a CDR CSV (cdr_csv.c), cdr_mysql.c, and I saw on the list a few weeks ago cdr_sybase.c What would help the Asterisk community the most? Is there really a demand out there for anything other than MySQL CSV flat file TDS (Sybase and MS SQL Server) I have never heard the words Oracle or Postgres thrown around so I am wondering if there is a justifiable need to support such systems. Yes for postgres. You will have a tough sell for the oracle route because of licensing unless there is a free interface available. I will do TDS. That is a given. However, before I go off and create something, I want the Asterisk community input as to what their needs are. Our needs: Asterisk to populate a CDR table in MS SQL Server Something that may need to be thought out, we seem to run into this database interface problem regularly enough. Is it time that the database access get moved to a resource and then the extensions app_sql_postgres.c could be made into just a sql app that works with any backend database. Also the CDR stuff could use the same interface to write out it's logs. Of course the biggest problem I see is that there isn't a fully set up perl DBI/DBD drivers for C yet. The libdbi library is still short a few critical drivers for what has been mentioned here, specifically CSV and MS SQL. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background messages while waiting for pick-up
Hi, we are a small office and would like * to answer the phone after 5 seconds and play a greeting. In parallel a phone should continue to ring. When the phone gets picked up, the call should be transfered. Is this possible? Any ideas/pointers are greatly appreciated. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c)or unixODBC (cdr_unixodbc.c) ?
Something that may need to be thought out, we seem to run into this database interface problem regularly enough. Is it time that the database access get moved to a resource and then the extensions app_sql_postgres.c could be made into just a sql app that works with any backend database. Also the CDR stuff could use the same interface to write out it's logs. Of course the biggest problem I see is that there isn't a fully set up perl DBI/DBD drivers for C yet. The libdbi library is still short a few critical drivers for what has been mentioned here, specifically CSV and MS SQL. Sorry [Steven Critchfield]. Are you saying to use libdbi versus unixODBC? I do not understand what your recommendation/suggestion is. unixODBC is probably just 1 means to an ends so multiple Databases can be used for CDR records. Erik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] phone rings while already on a call
Our office is set up with Budgetones internally. Occasionally, someone will be on the phone, and their phone will ring. How can I make it so that it will go straight to voicemail? Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk community input: FreeTDS(cdr_tds.c)or unixODBC (cdr_unixodbc.c) ?
On Fri, 2003-08-01 at 10:56, Erik Anderson wrote: Something that may need to be thought out, we seem to run into this database interface problem regularly enough. Is it time that the database access get moved to a resource and then the extensions app_sql_postgres.c could be made into just a sql app that works with any backend database. Also the CDR stuff could use the same interface to write out it's logs. Of course the biggest problem I see is that there isn't a fully set up perl DBI/DBD drivers for C yet. The libdbi library is still short a few critical drivers for what has been mentioned here, specifically CSV and MS SQL. Sorry [Steven Critchfield]. Are you saying to use libdbi versus unixODBC? I do not understand what your recommendation/suggestion is. I was actually saying in the last line quoted above that libdbi fell short of what we needed. unixODBC is probably just 1 means to an ends so multiple Databases can be used for CDR records. The benefits of something like libdbi is that it tries to use the native drivers first. You should be able to select a ODBC driver if no native driver exists. Native drivers should always be faster than an ODBC driver. My main point is that if we could consolidate all database activity into a centralized resource then all other parts of asterisk that people wish to have database support could then go to that resource and not worry about the underlying database. I think someone here mentioned having a mysql extension lookup module already working, this could then be modified to work with the resource and work on any supported database. Same with the CDR stuff. In general you should then be able to just drop a small amount of config for a module in a [general] section the defined the method of access and any pertinent authentication details and be done. I just wish something like perls DBI/DBD was available in native C and was as feature complete as perls implementation. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c)or unixODBC (cdr_unixodbc.c) ?
On 1 Aug 2003, Steven Critchfield wrote: Something that may need to be thought out, we seem to run into this database interface problem regularly enough. Is it time that the database access get moved to a resource and then the extensions app_sql_postgres.c could be made into just a sql app that works with any backend database. 1. Define an API providing whatever functions you might need. There probably aren't many - open DB, query, close DB, report last error, what else? Figure out which functions you need, and write prototypes for them. 2. For each type of database, write and compile the functions defined in step 1 into a shared library. 3. Allow the user to choose which library they need, and dl() it at runtime. -- JustThe.net Internet Multimedia Svcs. [The Fusion of Content Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor app
Thanks to Digium for the fine work they are doing on queue app! Does anyone know the progress on the monitor app recording to one file or synchronizing the two files' start and end times? Thanks. Jim Friedeck ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pcphoneline producs
Hi, Has anyone used http://www.pcphoneline.com/ products with asterisk? Senad J attachment: winmail.dat
Re: [Asterisk-Users] Mutex problem in sip?
It doesn't look like a problem. It's that when you have so many calls ... execution of some piece of code protected by mutex takes longer so it happens that some calls wait for their time . I guess if you have too many of those messages you should disable them. regards Martin On Thu, 31 Jul 2003, Alex Zarubin wrote: Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e Error -e eventually p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Got it eventually... . chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 948 (sip_hangup): Error obtaining mutex: Device or resource busy channel.c line 370 (ast_queue_frame): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy Thank you. Alex Zarubin Webley Systems, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor app
Hi, You can do this using wmux and sox from inside the Asterisk. Then get a single GSM file with the two streams mixed. Take a look at http://www.loligo.com/asterisk/ for an excelent example. Just replace toast with sox. It works like a charm for me. Best regards, Dan - Original Message - From: Jim Friedeck [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 9:32 PM Subject: [Asterisk-Users] Monitor app Thanks to Digium for the fine work they are doing on queue app! Does anyone know the progress on the monitor app recording to one file or synchronizing the two files' start and end times? Thanks. Jim Friedeck ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Overlap on PRI to PSTN
Hi, BellSouth provides us with a PRI, 6B+1D. Currently it is setup up to present incoming calls top-to-bottom (channel 6,5,4, etc) and expects outgoing in teh opposite order (1,2,3,etc). Incoming calls take priority and drop outgoing calls. DID is presented as 4 digits (even though our DID block spans 2 exchanges, go figure. Luckily not using the upper DID range). Would I configure 1 or 2 groups to support the dial-in and dial-out context? For outgoing calls, I assume I leave the Caller-ID set as asrecieved? Anything special needs doing in zaptel, Zapata, or other files? An example or pointers would be appreciated! Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phone rings while already on a call
Hi, I think that you must disable Call Waiting functionality. BR, Dan - Original Message - From: Steve Meyers [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Friday, August 01, 2003 7:09 PM Subject: [Asterisk-Users] phone rings while already on a call Our office is set up with Budgetones internally. Occasionally, someone will be on the phone, and their phone will ring. How can I make it so that it will go straight to voicemail? Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phone rings while already on a call
On Fri, 2003-08-01 at 13:50, Dan wrote: I think that you must disable Call Waiting functionality. I can't find where to disable it... I set callwaiting=no in zapata.conf and sip.conf, but neither seemed to help. I grepped for callwaiting in /etc/asterisk and couldn't find anything helpful. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seting up TDM40B
Hi list, I'm trying to set up a TDM40B, but modprobe returns the fowling errors: asterisk:~# modprobe wcfxs ZT_CHANCONFIG failed on channel 1: Invalid argument (22) /lib/modules/2.4.18/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.18/misc/wcfxs.o: insmod wcfxs failed Could someone give me a help? Thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seting up TDM40B
Title: RE: [Asterisk-Users] Seting up TDM40B Have you setup the zaptel.conf and zapata.conf configuration files for how ever many ports you have on the card and then run the ztcfg -vvvc command? -Original Message- From: Eduardo Goncalves [mailto:[EMAIL PROTECTED]] Sent: Friday, August 01, 2003 03:24 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Seting up TDM40B Hi list, I'm trying to set up a TDM40B, but modprobe returns the fowling errors: asterisk:~# modprobe wcfxs ZT_CHANCONFIG failed on channel 1: Invalid argument (22) /lib/modules/2.4.18/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.18/misc/wcfxs.o: insmod wcfxs failed Could someone give me a help? Thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
What does 'dmesg' say ? On Fri, 1 Aug 2003, Eduardo Goncalves wrote: Hi list, I'm trying to set up a TDM40B, but modprobe returns the fowling errors: asterisk:~# modprobe wcfxs ZT_CHANCONFIG failed on channel 1: Invalid argument (22) /lib/modules/2.4.18/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.18/misc/wcfxs.o: insmod wcfxs failed Could someone give me a help? Thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 15:25:49 -0500 McAughan, Matt [EMAIL PROTECTED] wrote: Have you setup the zaptel.conf and zapata.conf configuration files for how ever many ports you have on the card and then run the ztcfg -vvvc command? Since the module aren't loaded, config zaptel.conf, zapata.conf and run ztcfg will not work, right? my conf files: === zaptel.conf === fxsls=1-4 loadzone = us defaultzone=us === zapata.conf === callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 immediate=no context=pstn signalling=fxs_ks channel=1-4 thanks []'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seting up TDM40B
Title: RE: [Asterisk-Users] Seting up TDM40B Not sure about that one still an asterisk newbie myself but had similar problems. The only other thing I can suggest and as stupid as this might sound... is the card seated properly on the motherboard? -Original Message- From: Eduardo Goncalves [mailto:[EMAIL PROTECTED]] Sent: Friday, August 01, 2003 03:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Seting up TDM40B On Fri, 1 Aug 2003 15:25:49 -0500 McAughan, Matt [EMAIL PROTECTED] wrote: Have you setup the zaptel.conf and zapata.conf configuration files for how ever many ports you have on the card and then run the ztcfg -vvvc command? Since the module aren't loaded, config zaptel.conf, zapata.conf and run ztcfg will not work, right? my conf files: === zaptel.conf === fxsls=1-4 loadzone = us defaultzone=us === zapata.conf === callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 immediate=no context=pstn signalling=fxs_ks channel=1-4 thanks []'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seting up TDM40B
Title: RE: [Asterisk-Users] Seting up TDM40B And someone on this list can correct me if I am wrong but TDM40P is up to a 4 port FXS device so your signalling should be of one of the fxo derivitives. Right? -Original Message- From: Eduardo Goncalves [mailto:[EMAIL PROTECTED]] Sent: Friday, August 01, 2003 03:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Seting up TDM40B On Fri, 1 Aug 2003 15:25:49 -0500 McAughan, Matt [EMAIL PROTECTED] wrote: Have you setup the zaptel.conf and zapata.conf configuration files for how ever many ports you have on the card and then run the ztcfg -vvvc command? Since the module aren't loaded, config zaptel.conf, zapata.conf and run ztcfg will not work, right? my conf files: === zaptel.conf === fxsls=1-4 loadzone = us defaultzone=us === zapata.conf === callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 immediate=no context=pstn signalling=fxs_ks channel=1-4 thanks []'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 15:25:49 -0500 McAughan, Matt [EMAIL PROTECTED] wrote: Have you setup the zaptel.conf and zapata.conf configuration files for how ever many ports you have on the card and then run the ztcfg -vvvc command? Since the module aren't loaded, config zaptel.conf, zapata.conf and run ztcfg will not work, right? The modules are loading. Its just that the post-install ztcfg is erroring out. my conf files: === zaptel.conf === fxsls=1-4 This is why. FXS cards use FXO signalling vice versa. change this to fxols=1-4 loadzone = us defaultzone=us === zapata.conf === callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 immediate=no context=pstn signalling=fxs_ks This also needs to be changed to fxo. You've got a mismatch here as well. zaptel.conf is set to use loop-start signalling zapata.conf is expecting kewl-start signalling. I'd recommend changing zaptel.conf to fxoks and changing zapata.conf to signalling=fxo_ks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 15:34:23 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: What does 'dmesg' say ? CSLIP: code copyright 1989 Regents of the University of California PPP generic driver version 2.4.1 Zapata Telephony Interface Registered on major 196 Freshmaker version: 62 Freshmaker passed register test Module 0: Initialized Module 1: Initialized Module 2: Initialized Timeout waiting for calibration of module 3 ProSlic died on Calibration. Module 3: Not installed Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) and then the errors that I mentioned. Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy usb-ohci?
Is it possible to get ztdummy working with usb-ohci? Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco AS5300 -- Not hearing anything
Hi to all! I have this config, PSTN -- AS5300 -- ASTERISK I am using the Cisco as5300 to receive incoming calls and routing them to Asterisk for IVR. When I ran asterisk this is what I get when calling the voicemail demo. *CLI -- Executing Playback(SIP/-081058b8, transfer|skip) in new stack -- Executing Macro(SIP/-081058b8, stdexten|1234|Console/dsp) in new stack -- Executing Dial(SIP/-081058b8, Console/dsp|20) in new stack WARNING[1192437440]: File channel.c, Line 1558 (ast_request): No channel type registered for 'Console' NOTICE[1192437440]: File app_dial.c, Line 495 (dial_exec): Unable to create channel of type 'Console' == Everyone is busy at this time -- Executing VoiceMail(SIP/-081058b8, b1234) in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/1234/busy' -- Playing 'vm-intro' -- Playing 'beep' -- Recording to /var/spool/asterisk/vm/1234/INBOX/msg0015 -- User hung up == Parsing '/etc/asterisk/voicemail.conf': Found == Spawn extension (macro-stdexten, s, 102) exited non-zero on 'SIP/-081058b8' in macro 'stdexten' == Spawn extension (default, s, 2) exited non-zero on 'SIP/-081058b8' But in the phone I can't hear anything, I've tested also the voicemail with a software sip phone and It works great. But with the cisco I hear nothing, I'v tested codecs ulaw and alaw but the both do the same. This is my cisco's config dial-peer voice 20 voip destination-pattern 02322663910 translate-outgoing called 20 session protocol sipv2 session target ipv4:200.85.96.230 dtmf-relay cisco-rtp codec g711alaw ! translation-rule 20 Rule 0 ^02322663910 1234 ! Any ideas?? Luciano Ramos CCNA - MCP Jefe Depto. Internet TelViso 02320-470300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
Try to uncomment in zaptel/Makefile KFLAGS+=-DNO_CALIBRATION and make clean install that should help Martin On Fri, 1 Aug 2003, Eduardo Goncalves wrote: On Fri, 1 Aug 2003 15:34:23 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: What does 'dmesg' say ? CSLIP: code copyright 1989 Regents of the University of California PPP generic driver version 2.4.1 Zapata Telephony Interface Registered on major 196 Freshmaker version: 62 Freshmaker passed register test Module 0: Initialized Module 1: Initialized Module 2: Initialized Timeout waiting for calibration of module 3 ProSlic died on Calibration. Module 3: Not installed Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) and then the errors that I mentioned. Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP session traversing Asterisk server...
Ricardo, You are right about the contact field in the INVITE message. It does display the address or our Asterisk proxy. It seems to me that this field is used for endpoints to exchange future SIP messages among themselves and not to set up the RTP stream. I have found that the SDP Connection (c) field in the invite also reflects the IP of the Asterisk box after the message leaves the proxy. The 200 OK reflects the same symptoms. I think that this is the reason the RTP stream is being set up between the endpoint and the server. Do you think that the contact field and connection field being incorrect may be related? You also have mentioned that you have not seen a way to configure this with Asterisk. How about other SIP proxies such as VOCAL? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 2:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server... Dave, You can use a sniffer to view the contact field in the INVITE Message that the Originating Phone sends to *. Then look at the INVITE Message that * sends to the remote phone and compare the contact filed. You will see that the IP Address is changed to reflect the IP of *. If you want pure P2P then that address needs to remain the same. I have not seen how you can do that with *. Ricardo - Original Message - From: Dave Packham [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 3:00 PM Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server... OK calls thru the * server are looped and calls with the same phones thru Free WOrld Dialup are P2P. same configs... Anyone have any ideas? I know its a bug but we need to fix this one I think its pretty big one. it would HAMMER the scalability of * servers Dave [EMAIL PROTECTED] 7/29/2003 8:01:41 AM Sure, nothing special though: [4840] type=friend username=4840 host=dynamic canreinvite=yes nat=no qualify=200 mailbox=4840 dtmfmode=inband [4842] type=friend username=4842 host=dynamic canreinvite=yes nat=no qualify=200 mailbox=4840 dtmfmode=inband -Original Message- From: Dave Packham [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 15:43 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server ... can you share the SIP conf entries that you are using to get this to work? I have played with the canreinvite and reinvite entries but cannot make my 7960's do P2P I am running the 5.1 SIP code on the phones. Dave [EMAIL PROTECTED] 7/29/2003 3:13:54 AM Thanks all, I spent some time on this last night with packet sniffer in hand, the 'canreinvite' option makes sense and seems to work well for me (running latest * CVS release) when used between 79xx phones and the AS5300 gateway although I get some somewhat expected problems with 79xx that are NAT'd behind ADSL/cable connections. I don't seem to be hitting the bug that Dave mentioned below ... -Original Message- From: Dave Packham [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 its a know problem. I have played with the canreinvite stuff to no end and have never gotten my Cisco Phones to do P2P RTP. I am going to try free world dialup to see if it does P2P with my Cisco Phones then it might just be a message thing on * server. Dave Packham [EMAIL PROTECTED] 7/28/2003 4:16:16 PM On your sip.conf for each sip endopoint set canreinvite = yes. That way the rtp stream won t go through *. The only problem though is for ATA 186. They need canreinvite = No when they are in a NAT environment. - Original Message - From: Low, Adam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 28, 2003 11:29 AM Subject: [Asterisk-Users] RTP session traversing Asterisk server ... I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the server and ultimately a higher latency between the two end points. Is this a typical operation of Asterisk or is this possibly due to the fact that some of the phones (not those used in the tests) are running NAT and Asterisk relays all RTP packets ? Adam *
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 16:11:12 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Try to uncomment in zaptel/Makefile KFLAGS+=-DNO_CALIBRATION and make clean install that should help with this line uncommented, the module loads fine CSLIP: code copyright 1989 Regents of the University of California PPP generic driver version 2.4.1 Zapata Telephony Interface Registered on major 196 Freshmaker version: 62 Freshmaker passed register test Module 0: Initialized Module 1: Initialized Module 2: Initialized Module 3: Initialized Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) I also did the corrections on zapata and zaptel like James Sharp suggested. But when I try to run asterisk I get the fowling errors: == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1024]: File config.c, Line 537 (cfg_process): parse error: No category context for line 1 of zapata.conf ERROR[1024]: File chan_zap.c, Line 6355 (load_module): Unable to load config zapata.conf WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! :~ Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 16:31:36 -0500 (CDT) James Sharp [EMAIL PROTECTED] wrote: == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1024]: File config.c, Line 537 (cfg_process): parse error: No category context for line 1 of zapata.conf zapata.conf needs to start with the line [channels] My mistake. I had commented this line when I did the regex for uncomment the others. Sorry, and thanks for the help :~ [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP session traversing Asterisk server...
Hi Andrew, After looking at some SIP messages again I too think the (c) field in the SDP is what determines the RTP endpoints. It's just that in our case it is always the same as the Contact field. In any case what you see here is that * is making some changes here to make sure SIP messages and RTP stream passes through it. If what you want is a plain but powerful SIP Proxy then take a look at (SER) http://www.iptel.org. That is what we use to run our SIP P2P network. We only use * for our PBX. Regards, Ricardo http://www.telesip.net - Original Message - From: Andrew Reich [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 4:20 PM Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server... Ricardo, You are right about the contact field in the INVITE message. It does display the address or our Asterisk proxy. It seems to me that this field is used for endpoints to exchange future SIP messages among themselves and not to set up the RTP stream. I have found that the SDP Connection (c) field in the invite also reflects the IP of the Asterisk box after the message leaves the proxy. The 200 OK reflects the same symptoms. I think that this is the reason the RTP stream is being set up between the endpoint and the server. Do you think that the contact field and connection field being incorrect may be related? You also have mentioned that you have not seen a way to configure this with Asterisk. How about other SIP proxies such as VOCAL? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 2:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server... Dave, You can use a sniffer to view the contact field in the INVITE Message that the Originating Phone sends to *. Then look at the INVITE Message that * sends to the remote phone and compare the contact filed. You will see that the IP Address is changed to reflect the IP of *. If you want pure P2P then that address needs to remain the same. I have not seen how you can do that with *. Ricardo - Original Message - From: Dave Packham [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 3:00 PM Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server... OK calls thru the * server are looped and calls with the same phones thru Free WOrld Dialup are P2P. same configs... Anyone have any ideas? I know its a bug but we need to fix this one I think its pretty big one. it would HAMMER the scalability of * servers Dave [EMAIL PROTECTED] 7/29/2003 8:01:41 AM Sure, nothing special though: [4840] type=friend username=4840 host=dynamic canreinvite=yes nat=no qualify=200 mailbox=4840 dtmfmode=inband [4842] type=friend username=4842 host=dynamic canreinvite=yes nat=no qualify=200 mailbox=4840 dtmfmode=inband -Original Message- From: Dave Packham [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 15:43 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server ... can you share the SIP conf entries that you are using to get this to work? I have played with the canreinvite and reinvite entries but cannot make my 7960's do P2P I am running the 5.1 SIP code on the phones. Dave [EMAIL PROTECTED] 7/29/2003 3:13:54 AM Thanks all, I spent some time on this last night with packet sniffer in hand, the 'canreinvite' option makes sense and seems to work well for me (running latest * CVS release) when used between 79xx phones and the AS5300 gateway although I get some somewhat expected problems with 79xx that are NAT'd behind ADSL/cable connections. I don't seem to be hitting the bug that Dave mentioned below ... -Original Message- From: Dave Packham [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 its a know problem. I have played with the canreinvite stuff to no end and have never gotten my Cisco Phones to do P2P RTP. I am going to try free world dialup to see if it does P2P with my Cisco Phones then it might just be a message thing on * server. Dave Packham [EMAIL PROTECTED] 7/28/2003 4:16:16 PM On your sip.conf for each sip endopoint set canreinvite = yes. That way the rtp stream won t go through *. The only problem though is for ATA 186. They need canreinvite = No when they are in a NAT environment. - Original Message - From: Low, Adam [EMAIL PROTECTED]
[Asterisk-Users] Using OH323 and Gatekeeper
Hello all, Please forgive me if this sounds a little (or a lot) ignorant as I am very new to asterisk. Right now I have two pc's connected back to back through an E100 card running asterisk. I have openh323 running as well and I am able to route calls through the E1 line. Next up I would like to be able to register asterisk with a gatekeeper. On another computer is running openGK. Using the gatekeeper I am able to register my h323 client with a phone number, it all seems to be working. When I put the ip address of the gatekeeper into h323.conf, asterisk failed to load. I get an error that I can't open an h323 listener on port 1720. Has anyone done this before? Your help would be appreciated. Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callwaiting in sip can't be disabled
At least any way I've tried. I put callwaiting = no in sip.conf in the [general] section and in the section for my specific phone, and it still sends through calls even though I'm already on the line. How can I disable it? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP!!!!
Hi When i make a call using oh323 is posible to make the ringing sound, to indicate the progress of the call At this time when i make a call, i can't hear anything unless time out or the call is made Please hellp!! thanks Sorry for my english ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using OH323 and Gatekeeper
You shouldn't run the gatekeeper and asterisk on the same machine. Jeremy McNamara Langley, Sean wrote: Hello all, Please forgive me if this sounds a little (or a lot) ignorant as I am very new to asterisk. Right now I have two pc's connected back to back through an E100 card running asterisk. I have openh323 running as well and I am able to route calls through the E1 line. Next up I would like to be able to register asterisk with a gatekeeper. On another computer is running openGK. Using the gatekeeper I am able to register my h323 client with a phone number, it all seems to be working. When I put the ip address of the gatekeeper into h323.conf, asterisk failed to load. I get an error that I can't open an h323 listener on port 1720. Has anyone done this before? Your help would be appreciated. Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using OH323 and Gatekeeper
Langley, Sean wrote: When I put the ip address of the gatekeeper into h323.conf, asterisk failed to load. I get an error that I can't open an h323 listener on port 1720. Something is already using port 1720. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP!!!! Ringback oh323
Hi What command i need to use to make a call with oh323 and hear the ringback sound Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!! Ringback oh323
Specify option 'r' to dial application. Michael On Friday 01 August 2003 07:13 pm, [EMAIL PROTECTED] wrote: Hi What command i need to use to make a call with oh323 and hear the ringback sound Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seting up TDM40B
I had the same exact problem yesterday Eduardo. Also make sure that you provide the -c flag to ztcfg. I didn't have to recompile any pieces of *, you shouldn't either. ztcfg -c /etc/asterisk/zaptel.conf -vvv will do the trick once you get the files, signalling (e.g. fxo_ko) and good configs. Check back one day on the list for my followup. --- Gavin -Original Message- From: Eduardo Goncalves [mailto:[EMAIL PROTECTED] Sent: Fri 8/1/2003 5:54 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] Seting up TDM40B On Fri, 1 Aug 2003 16:31:36 -0500 (CDT) James Sharp [EMAIL PROTECTED] wrote: == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1024]: File config.c, Line 537 (cfg_process): parse error: No category context for line 1 of zapata.conf zapata.conf needs to start with the line [channels] My mistake. I had commented this line when I did the regex for uncomment the others. Sorry, and thanks for the help :~ [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users [EMAIL PROTECTED])fjåËbú?jË^®+$º
[Asterisk-Users] Hangup after a Timeout
hi everybody, can anybody pls tell me a way to hangup an answered Zap/SIP/IAX call, after a specified time period expires, like after 10, 15 minutes. Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected
Then I am afraid that i can't help much.May be someone else will. Rgds Manoj K Gupta - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 12:44 PM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Hi, It is the same with any other application I try to run from the System() application. I don't think is a privilege problem. Even with 'ls' as command, it displays the directory listing and then exit with error. Thanks, Dan - Original Message - From: Sip Rtp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 10:07 AM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Does, it not look like a privilege problem on the file /usr/local/bin/wmix? Please check that also. Rgds SIP - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 31, 2003 10:30 PM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Yes. It works even in the macro (not only oin the shell), but because of that warning, the macro exit. The resulting file of the mix is perfect (tried after that), but still that warning. See my previous mail ... testet with a simple 'ls' command. I don't know what to do...:( BR, Dan - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Thursday, July 31, 2003 7:49 PM Subject: Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected Try to do the same in shell. Does it work ? Martin On Thu, 31 Jul 2003, Dan wrote: Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple wmix without any parameter and I get the same error. . -- Executing System(SIP/103-b7c0, /usr/local/bin/wmix /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav /var/spool/asterisk/monitor/31072003-19:08:11-103.wav) in new stack WARNING[1200825920]: File app_system.c, Line 57 (skel_exec): Unable to execute '/usr/local/bin/wmix /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav /var/spool/asterisk/monitor/31072003-19:08:11-103.wav' == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on 'SIP/103-b7c0' in macro 'record-cleanup' == Spawn extension (fullaccess, h, 1) exited non-zero on 'SIP/103-b7c0' Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
