Re: [Asterisk-Users] Asterisk/Zaptel 64-bit?
On Tue, 2006-05-09 at 02:03 +0800, Kenige Ho wrote: Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries?I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406 Digium Card OS: Fedore Core 5Although I don't have any experience with your particular hardware,afaik there is no issue with building and running zaptel, libpri and asterisk on x86_64 hardware. Works fine for many. You can find (S)RPMsat http://atrpms.net and at http://www.laimbock.com/asterisk/ If you search on http://voip-info.org there are more places where you can findpackages for FC5. If I may make a suggestion: have you thought aboutgetting another harddisk and run the two in a RAID1 setup? The cost of the extra harddisk versus the cost of being down on a 4 span card issomething probably worth considering.Regards,PatrickThank for your reply and advice, Patrick. The reason I didn't add a RAID1 to the server was that I am not sure if RAID1 on SATA II is stable on FC5 yet and also since the HDD isn't hotswap (at least i don't think SATA HDD can hotswap). There will always be a downtime for me. And future expansion will include a standalone mySQL DB for Realtime Asterisk in which many Asterisk server will point to this mySQL server and get sip users and write CDRs too. But this is all testing for now. As I will also need to to deploy a SER to make this all work. But the funny thing is that most of the hardware now it 64-bit, I would need to find high and low to make sure that the HCL is good with FC5. Also for the zaptel driver and sources, I prefer to a compile on it rather than rpm install. So I wanted to make sure that I wouldn't encounter any problems with a compile and don't want to waste my hardware purchase. Is there any special command or switches that I need to add in the make files to compile to 64-bit? Thank you, Patrick and thanks to everyone that has taken the time to read my previous email. Regards,Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk/Zaptel 64-bit?
Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406 Digium Card OS: Fedore Core 5 Thank you for your input. Regards, Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fake Ring Tone/Compile Addon
Subject: Re: [Asterisk-Users] Fake Ring Tone/Compile AddonTo: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO-8859-1; format=flowed Kenige Ho wrote: Dear All, I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem?Don't use the fake ring option to dial. This is the r option. Dear Manxpower, I didn't use the 'r' option in my Dial command, and the funny thing that out going to my SIP Phones doesn't have fake ring tone. But there is always fake ring tone, when sending out to the VoIP Gateway and I am sure that I don't set it in the VoIP gateway. Please help. Thanks. Regards, Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fake Ring Tone/Compile Addon
Dear All,I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? This problem is existing in my Asterisk 1.2.1 box. Also when compiling Asterisk 1.2.5 and tried to run it with the Asterisk-addon, the CLI output would get segmentation default. The source of the problem was that res_config_mysql.so was giving problem. It was when compiling the asterisk-addon, it was using the default Libarary and includes of the Fedore Core mysql directories. But my box is now running MySQL 5, and the source directory is at /root/mysql- standard-5.0.16-linux-i686 directory. My question is how to compile using this new source code for the header files and the libarary files? Many thanks.Regards,Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ooh323 Gatekeeper Bug
Dear All,It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a gatekeeper. This current version i am using for the ooh323 is from Asterisk-addon-1.2.1. Is there a bug on this version for the ooh323 and also how can i get the newer version of the ooh323(0.8.1) to compile with? Many thanks to you all.Regards,Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP-H323 Help and Multiple Listening Port
Dear All, I have a very strange situation here and wondering if anyone can assist me. I am trying to connect an H323 call from an GnuGK to Asterisk 1.2.1 which routes the call to an SIP Hard Phone. The funny thing that I can collect the connect but the call always drop about 1 second or 2 seconds after it is connect. I am not sure if this will help but I do see some 'Trapped RCF' in the GnuGK logs. Many thanks. Diagram of Network: External GK(Public IP)-GnuGK(Public IP)-Asterisk(Public IP)-SIP Phone(NATed) Also I would like to ask if it is possible for Asterisk to be listening on 5060 and another non-standard at the same time? Thanks. Regards, Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 18, Issue 158
Hi, I have already set canreinvite=no in the sip.conf and also used the NAT=yes. But the funny thing that was in one case the user call and it wasn't working (one way audio as described) using an online dialer and then tried again using X-lite it was working. Then hanged up and tried X-lite again, it was not working. The second call was only a few seconds apart. Moving back to the online dialer, it wasn't working either. So it is just very strange to me how this happened and i was think maybe it was the RTP negiotation. Do you have any ideas? Regards, KengieFew people, or no one, will take the time to see all the debug.The key here is that the RTP port and IP negotiated in the SDP messagesent by asterisk to each party, should be visible for the party. A common error is Asterisk sending in SDP a private IP address to apublic UA, so the public UA will attempt to send RTP audio to aprivate IP, never reaching the Asterisk Server. Check voip-info.orgabout RTP issues with NAT, check the option canreinvite in sip.conf,put canreinvite=no , may be that will help. If you have one of the UAbehind a NAT, use nat=yesregards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP RTP Negotiation
HiAsterisk-Users, Please help as it is out of my league and understand as why the call would be silent or partly silent (calling party can't hear the called party, but called party can hear). Thanks in advance. Regards, Kengie On 1/19/06, Kenige Ho [EMAIL PROTECTED] wrote: Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 1 to 2 UA Listen RTP Port is 15000 Below is the the SIP Logs TestServer*CLI -- SIP read from 66.193.155.2:46478: REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0 Via: SIP/2.0/UDP 172.28.174.25:5060 From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERContact: XX sip:[EMAIL PROTECTED]:5060User-Agent: X UserAgent/1.0Expires: 5000Max-Forwards: 70Content-Length: 0 --- (11 headers 0 lines)---Using latest REGISTER request as basis requestSending to 172.28.174.25 : 5060 (non-NAT) Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 100 TryingVia: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ---Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 172.28.174.25:5060;received= 66.193.155.2From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED];tag=as504de7b8 Call-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=31e7aa76 Content-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms TestServer*CLI -- SIP read from 66.193.155.2:46478: REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0 Via: SIP/2.0/UDP 172.28.174.25:5060From: XX sip:[EMAIL PROTECTED] To: XX sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTER Contact: XX sip:[EMAIL PROTECTED]:5060Authorization: Digest username=XX, realm=asterisk, nonce=31e7aa76, uri=sip: XXX.XXX.XX.XXX, response=3e54ea5e3a3b6df1e5db7b4a3182e18f, algorithm=MD5 User-Agent: X UserAgent/1.0Expires: 5000Max-Forwards: 70Content-Length: 0 --- (12 headers 0 lines)---Using latest REGISTER request as basis requestSending to 172.28.174.25 : 5060 (NAT) Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 100 TryingVia: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ---Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 200 OKVia: SIP/2.0/UDP 172.28.174.25:5060;received= 66.193.155.2From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED];tag=as504de7b8 Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Expires: 3600Contact: sip:[EMAIL PROTECTED]:5060;expires=3600Date: Wed, 18 Jan 2006 11:20:15 GMTContent-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]'TestServer*CLI -- SIP read from 202.83.167.103:30928: --- (0 headers 0 lines) Nat keepalive ---TestServer*CLI -- SIP read from 218.111.26.5:5060 : --- (0 headers 0 lines) Nat keepalive ---Destroying call '[EMAIL PROTECTED]' TestServer*CLI -- SIP read from 66.193.155.2:46478: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.28.174.25:5060From: XX sip:[EMAIL PROTECTED] ;tag=174cf52To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 3 INVITEContact: sip:[EMAIL PROTECTED] :5060Subject: no subjectUser-Agent: X UserAgent/1.0Max-Forwards: 70Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTERContent-Type: application/SDP Accept: application/SDP, text/plain Accept-Encoding: identityContent-Length: 287 v=0o=XX 11375833482 11375833482 IN IP4 172.28.174.25s=VaxSoft Inc.c=IN IP4 172.28.174.25t=0 0m=audio 15000 RTP/AVP 3 98 8 0 101 a=rtpmap:3 GSM/8000a=rtpmap:98 iLBC/8000a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16 --- (15 headers 12 lines)---Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 172.28.174.25 : 5060 (non-NAT)Reliably Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 172.28.174.25:5060 ;received= 66.193.155.2From: XX sip:[EMAIL PROTECTED];tag=174cf52To: sip
[Asterisk-Users] SIP RTP Negotiation
Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 1 to 2 UA Listen RTP Port is 15000 Below is the the SIP Logs TestServer*CLI -- SIP read from 66.193.155.2:46478: REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0Via: SIP/2.0/UDP 172.28.174.25:5060 From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERContact: XX sip:[EMAIL PROTECTED]:5060User-Agent: X UserAgent/1.0Expires: 5000Max-Forwards: 70Content-Length: 0 --- (11 headers 0 lines)---Using latest REGISTER request as basis requestSending to 172.28.174.25 : 5060 (non-NAT)Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 100 TryingVia: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ---Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 172.28.174.25:5060;received= 66.193.155.2From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED];tag=as504de7b8 Call-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=31e7aa76Content-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 msTestServer*CLI -- SIP read from 66.193.155.2:46478: REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0Via: SIP/2.0/UDP 172.28.174.25:5060From: XX sip:[EMAIL PROTECTED] To: XX sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTER Contact: XX sip:[EMAIL PROTECTED]:5060Authorization: Digest username=XX, realm=asterisk, nonce=31e7aa76, uri=sip:XXX.XXX.XX.XXX, response=3e54ea5e3a3b6df1e5db7b4a3182e18f, algorithm=MD5 User-Agent: X UserAgent/1.0Expires: 5000Max-Forwards: 70Content-Length: 0 --- (12 headers 0 lines)---Using latest REGISTER request as basis requestSending to 172.28.174.25 : 5060 (NAT)Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 100 TryingVia: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ---Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 200 OKVia: SIP/2.0/UDP 172.28.174.25:5060;received= 66.193.155.2From: XX sip:[EMAIL PROTECTED]To: XX sip:[EMAIL PROTECTED];tag=as504de7b8 Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Expires: 3600Contact: sip:[EMAIL PROTECTED]:5060;expires=3600Date: Wed, 18 Jan 2006 11:20:15 GMTContent-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 msDestroying call '[EMAIL PROTECTED]'TestServer*CLI -- SIP read from 202.83.167.103:30928: --- (0 headers 0 lines) Nat keepalive ---TestServer*CLI -- SIP read from 218.111.26.5:5060: --- (0 headers 0 lines) Nat keepalive ---Destroying call '[EMAIL PROTECTED]'TestServer*CLI -- SIP read from 66.193.155.2:46478: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0Via: SIP/2.0/UDP 172.28.174.25:5060From: XX sip:[EMAIL PROTECTED] ;tag=174cf52To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 3 INVITEContact: sip:[EMAIL PROTECTED]:5060Subject: no subjectUser-Agent: X UserAgent/1.0Max-Forwards: 70Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTERContent-Type: application/SDP Accept: application/SDP, text/plainAccept-Encoding: identityContent-Length: 287 v=0o=XX 11375833482 11375833482 IN IP4 172.28.174.25s=VaxSoft Inc.c=IN IP4 172.28.174.25t=0 0m=audio 15000 RTP/AVP 3 98 8 0 101 a=rtpmap:3 GSM/8000a=rtpmap:98 iLBC/8000a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16 --- (15 headers 12 lines)---Using INVITE request as basis request - [EMAIL PROTECTED]Sending to 172.28.174.25 : 5060 (non-NAT)Reliably Transmitting (NAT) to 66.193.155.2:46478:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 172.28.174.25:5060 ;received=66.193.155.2From: XX sip:[EMAIL PROTECTED];tag=174cf52To: sip:[EMAIL PROTECTED];tag=as733e9adcCall-ID: [EMAIL PROTECTED]CSeq: 3 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm=asterisk, nonce=452293c4Content-Length: 0 ---Scheduling destruction
[Asterisk-Users] SIP Error 401 Problem
Dear All, I am having this strange problem on my Asterisk 1.2.1. We have a web dialer that can register to the Asterisk box in Hong Kong, but another user using the same account can't register to the Asterisk box using the same web dialer. Below is an output of the sip debug logs. It seems that the digest is missing the username and password, but why? I have also have this call flow for the an IP Phone, but after a while, it will register to the Asterisk. One thing I don't understand is that I have registered successfully in Hong Kong and when the user tries in South Africa, it doesn't work. Please Help! SIP Logs: From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERContact: *User-Agent: VaxSIP UserAgent/1.0Expires: 0Max-Forwards: 70Content-Length: 0 --- (11 headers 0 lines)---Using latest REGISTER request as basis requestSending to 192.168.0.3 : 2232 (non-NAT)Transmitting (NAT) to 196.38.228.123:5060:SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.0.3:2232;received=196.38.228.123From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ---Transmitting (NAT) to 196.38.228.123:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.0.3:2232;received= 196.38.228.123From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED];tag=as63889026 Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=4929aec7Content-Length: 0 Regards, Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AGI GET Variable Problem
Dear All,
Never Mind, I have solved the problem. It seems that you should clear the buffer for any 'waiting' response or else you will be getting an empty '200 result=1' response. So be sure to read, before you write in php agi script to ensure that you will get a proper response.
Regards,
Kengie
On 12/13/05, Kenige Ho [EMAIL PROTECTED] wrote:
Dear All,
I am trying to get a variable via AGI GET VARIABLE , but using AGI DEBUG I actually do see the variable get return but somehow my retrieving the variable via php. I don't get the value of the variable. Below is my code and my results. Please help. thank you.
Coding:
#!/usr/bin/php -q?phpob_implicit_flush(true);set_time_limit(6);$in = fopen(php://stdin,r);$stdlog = fopen(/var/log/asterisk/my_agi.log, w);
// toggle debugging output (more verbose)$debug = false;
// Do function definitions before we start the main loopfunction read() { global $in, $debug; $input = str_replace(\n, , fgets($in, 4096)); if ($debug) fputs($stdlog, read: $input\n);
return $input;}
function errlog($line) { global $err; echo VERBOSE \$line\\n;}
function write($line) { global $debug; if ($debug) fputs($stdlog, write: $line\n); echo $line.\n;}
// parse agi headers into arraywhile ($env=read()) { $s = split(: ,$env); // $agivar[str_replace(agi_,,$s[0])] = trim($s[1]); // errlog($s[0].,.$s[1]);
$agivar[$s[0]] = trim($s[1]); if(($endid.phpv==) || ($env==\n)) { break; }}
// main programecho VERBOSE \fone-check\\n;$tmp = GET VARIABLE x;write($tmp);errlog(Temp Var is . $tmp);$result = read();errlog(Before Strip Result is . $result);
$result = trim(ereg_replace(200 result=1,,$result));$result = trim(ereg_replace(\(,,$result));$result = trim(ereg_replace(\),,$result));
errlog(After Strip Result is . $result);
// clean up file handlers etc.fclose($in);fclose($stdlog);
exit;?
Results:
AGI Debugging EnabledAGI Tx agi_request: fone-check.agiAGI Tx agi_channel: SIP/1234-addaAGI Tx agi_language: enAGI Tx agi_type: SIPAGI Tx agi_uniqueid:
1134460079.22AGI Tx agi_callerid: 1234AGI Tx agi_calleridname: 1234AGI Tx agi_callingpres: 0AGI Tx agi_callingani2: 0AGI Tx agi_callington: 0AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 1233AGI Tx agi_rdnis: unknownAGI Tx agi_context: testAGI Tx agi_extension: 1233AGI Tx agi_priority: 11AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode: testAGI Tx AGI Rx VERBOSE fone-check fone-check.agi: fone-checkAGI Tx 200 result=1AGI Rx GET VARIABLE foneAGI Tx 200 result=1 (55)
AGI Rx VERBOSE Temp Var is GET VARIABLE fone fone-check.agi: Temp Var is GET VARIABLE foneAGI Tx 200 result=1AGI Rx VERBOSE Before Strip Result is 200 result=1
fone-check.agi: Before Strip Result is 200 result=1AGI Tx 200 result=1AGI Rx VERBOSE After Strip Result is fone-check.agi: After Strip Result is AGI Tx 200 result=1
Regards,
Kengie
___
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[Asterisk-Users] AGI GET Variable Problem
Dear All,
I am trying to get a variable via AGI GET VARIABLE , but using AGI DEBUG I actually do see the variable get return but somehow my retrieving the variable via php. I don't get the value of the variable. Below is my code and my results. Please help. thank you.
Coding:
#!/usr/bin/php -q?phpob_implicit_flush(true);set_time_limit(6);$in = fopen(php://stdin,r);$stdlog = fopen(/var/log/asterisk/my_agi.log, w);
// toggle debugging output (more verbose)$debug = false;
// Do function definitions before we start the main loopfunction read() { global $in, $debug; $input = str_replace(\n, , fgets($in, 4096)); if ($debug) fputs($stdlog, read: $input\n);
return $input;}
function errlog($line) { global $err; echo VERBOSE \$line\\n;}
function write($line) { global $debug; if ($debug) fputs($stdlog, write: $line\n); echo $line.\n;}
// parse agi headers into arraywhile ($env=read()) { $s = split(: ,$env); // $agivar[str_replace(agi_,,$s[0])] = trim($s[1]); // errlog($s[0].,.$s[1]);
$agivar[$s[0]] = trim($s[1]); if(($endid.phpv==) || ($env==\n)) { break; }}
// main programecho VERBOSE \fone-check\\n;$tmp = GET VARIABLE x;write($tmp);errlog(Temp Var is . $tmp);$result = read();errlog(Before Strip Result is . $result);
$result = trim(ereg_replace(200 result=1,,$result));$result = trim(ereg_replace(\(,,$result));$result = trim(ereg_replace(\),,$result));
errlog(After Strip Result is . $result);
// clean up file handlers etc.fclose($in);fclose($stdlog);
exit;?
Results:
AGI Debugging EnabledAGI Tx agi_request: fone-check.agiAGI Tx agi_channel: SIP/1234-addaAGI Tx agi_language: enAGI Tx agi_type: SIPAGI Tx agi_uniqueid:
1134460079.22AGI Tx agi_callerid: 1234AGI Tx agi_calleridname: 1234AGI Tx agi_callingpres: 0AGI Tx agi_callingani2: 0AGI Tx agi_callington: 0AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 1233AGI Tx agi_rdnis: unknownAGI Tx agi_context: testAGI Tx agi_extension: 1233AGI Tx agi_priority: 11AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode: testAGI Tx AGI Rx VERBOSE fone-check fone-check.agi: fone-checkAGI Tx 200 result=1AGI Rx GET VARIABLE foneAGI Tx 200 result=1 (55)
AGI Rx VERBOSE Temp Var is GET VARIABLE fone fone-check.agi: Temp Var is GET VARIABLE foneAGI Tx 200 result=1AGI Rx VERBOSE Before Strip Result is 200 result=1
fone-check.agi: Before Strip Result is 200 result=1AGI Tx 200 result=1AGI Rx VERBOSE After Strip Result is fone-check.agi: After Strip Result is AGI Tx 200 result=1
Regards,
Kengie
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[Asterisk-Users] Re: Marco and Realtime Extension Problem [SOLVED]
Dear All,
Sorry to be posting again. I have solved my problem.
The problem is that when exiting from the macro, the priority number
is still in effect. For example, priority 1 is at the start before
entering macro after the macro the priorty will be 2. Since there
isn't any other dialplan command, the switch statement would be search
for a priority 2 in the Realtime extensions table.
One Tip: If there isn't any one to help you, just help yourself. In
the logger.conf, use the console statement and output ready to the
console and you will be able to almost see everything that is going
on.
I want to thank the person that left that tip in the mailing list.
Sorry I forgot who it was as I was searching through the entire
archive from the begin.
I hope that this will help some people when there isn't any one to help you.
Regards,
Kengie Ho
On 7/21/05, Kenige Ho [EMAIL PROTECTED] wrote:
Dear All,
I have a problem with the Marco and the Realtime Extensions in my
extensions.conf. The problem is that when I exit from my Marco, I
should return to my calling context, which is default but the next
step for it should be switch statement which will use realtime
extension. Somehow I am getting the following error below with
autofallthrough=yes :
-- Executing NoOp(SIP/555-5dcf, Channel is SIP/555-5dcf) in new stack
== Auto fallthrough, channel 'SIP/555-5dcf' status is 'UNKNOWN'
And the following error with autofallthrough=no :
-- Executing NoOp(SIP/555-f121, Channel is SIP/555-f121) in new stack
Jul 21 16:51:46 WARNING[4090]: pbx.c:2337 __ast_pbx_run: Timeout, but
no rule 't' in context 'default'
In a sense, when I leave the marco, I should be able to enter the
realtime extension, but the call flow just fails after prority of the
default context.
Is there some bug in my sytnax or something in the asterisk program itself?
Below is my default context:
[default]
exten = _X.,1,Macro(stdexten,${EXTEN},${CALLERIDNUM})
;Realtime Routing from MySQL
switch = Realtime/[EMAIL PROTECTED]
[macro-stdexten]
exten = s,1,NoOp(Leaving Marco)
Regards,
Kengie Ho
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[Asterisk-Users] Marco and Realtime Extension Problem
Dear All,
I have a problem with the Marco and the Realtime Extensions in my
extensions.conf. The problem is that when I exit from my Marco, I
should return to my calling context, which is default but the next
step for it should be switch statement which will use realtime
extension. Somehow I am getting the following error below with
autofallthrough=yes :
-- Executing NoOp(SIP/555-5dcf, Channel is SIP/555-5dcf) in new stack
== Auto fallthrough, channel 'SIP/555-5dcf' status is 'UNKNOWN'
And the following error with autofallthrough=no :
-- Executing NoOp(SIP/555-f121, Channel is SIP/555-f121) in new stack
Jul 21 16:51:46 WARNING[4090]: pbx.c:2337 __ast_pbx_run: Timeout, but
no rule 't' in context 'default'
In a sense, when I leave the marco, I should be able to enter the
realtime extension, but the call flow just fails after prority of the
default context.
Is there some bug in my sytnax or something in the asterisk program itself?
Below is my default context:
[default]
exten = _X.,1,Macro(stdexten,${EXTEN},${CALLERIDNUM})
;Realtime Routing from MySQL
switch = Realtime/[EMAIL PROTECTED]
[macro-stdexten]
exten = s,1,NoOp(Leaving Marco)
Regards,
Kengie Ho
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