[Asterisk-Users] SIP channels not cleared
Hello all, When I do 'sip show channels' I have seen a lot of entries where these calls has already been terminated. Some of these channels are bolong to calls being made 2 days ago but still showing from the CLI. They look like 10.223.51.1730022676583 130b36625fc 00102/00103 unknow(d) Rx: BYE 10.223.51.1730022676583 5533069e578 00102/00103 unknow(d) Rx: BYE 10.223.51.1730016513973 234f7bba140 00102/00103 unknow(d) Rx: BYE 10.223.51.1730027226765 487b770b231 00102/00103 unknow(d) Rx: BYE 10.223.51.1730016513973 69b59aa2084 00102/00103 unknow(d) Rx: BYE 10.223.51.1730199820127 60ef984904a 00102/00103 unknow(d) Rx: BYE 10.223.51.1730081805135 45bf3e8c287 00102/00103 unknow(d) Rx: BYE I have thousands of them in 'sip show channels' and is increasing but it only shows 50 calls in 'show channels'. I believe this eats up memory. Sooner or later my system will run out of memory or get the 'Too many file opened' error. I have made a sip trace on asterisk and seems like they all share a same SIP message flow. When asterisk send an INVITE to other sip server say B. B will reply with Trying. When B found out that the actual destination can not be reached, it sends a BYE to asterisk. Asterisk then reply with a 200 OK. Call is hangup succesfully but 'sip show channels' still list the call record and never go away untill asterisk is restart. See below: Aug 15 18:35:32 VERBOSE[12402] logger.c: Reliably Transmitting (no NAT) to 10.223.51.173:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0^M Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4^M From: DADAS sip:[EMAIL PROTECTED];tag=as64c4813c^M To: sip:[EMAIL PROTECTED]^M Contact: sip:[EMAIL PROTECTED]^M Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Date: Mon, 15 Aug 2005 10:35:32 GMT^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY^M Content-Type: application/sdp^M Content-Length: 160^M ^M v=0^M o=root 12402 12402 IN IP4 10.21.99.221^M s=session^M c=IN IP4 10.21.99.221^M t=0 0^M m=audio 10986 RTP/AVP 8^M a=rtpmap:8 PCMA/8000^M a=silenceSupp:off - - - -^M Aug 15 18:35:32 VERBOSE[15229] logger.c: -- SIP read from 10.223.51.173:5060: SIP/2.0 100 Trying Call-Id: [EMAIL PROTECTED] CSeq: 102 INVITE From: DADAS sip:[EMAIL PROTECTED];tag=as64c4813c To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4 Aug 15 18:35:39 VERBOSE[15229] logger.c: -- SIP read from 10.223.51.173:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Call-Id: [EMAIL PROTECTED] Content-Length: 0 CSeq: 103 BYE From: sip:[EMAIL PROTECTED];tag=a10111834662596 To: DADAS sip:[EMAIL PROTECTED];tag=as64c4813c Via: SIP/2.0/UDP 10.223.51.173;branch=z9hG4bK05f6ab33 Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4 Aug 15 18:35:39 VERBOSE[15229] logger.c: Transmitting (no NAT) to 10.223.51.173:5060: SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.223.51.173;branch=z9hG4bK05f6ab33^M Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4^M From: sip:[EMAIL PROTECTED];tag=a10111834662596^M To: DADAS sip:[EMAIL PROTECTED];tag=as64c4813c^M Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY^M Contact: sip:[EMAIL PROTECTED]^M Content-Length: 0^M The SIP message exchange seems to be comply to the standard. Is this a bug in asterisk? I have a system where there is always call going on and I cant schedule asterisk to be restarted at any time to clear the channels. Any idea? I have CVS HEAD runnung on fedora 3. Thanks CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting Caller ID after Dial
Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Caller ID after Dial
Hey Mark, Have you tested on doing transfer (blind and attended)? Are the extensions in the CDR still correct? CCF --- Mark Johnson [EMAIL PROTECTED] wrote: Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF I forgot in my last post to mention that I use Postgres for my CDR, and the SIP extension can be pulled from the channel column. That way, the callerid is still the way it appeared when the calls were placed. I just strip everything from the '-' to the right and it's worked great for me! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Caller ID after Dial
Actually I already using account code for billing, so billing is fine. I have a 3rd party reporting software that tie extension numbers to departments. At the end of a month the person in charge will generate report on the call statistic for each department. My problem is now the report showing only one department are making calls because every outgoing call is from the same caller number. --- Chris A. Icide [EMAIL PROTECTED] wrote: What about setting and using Accountcode for each sip client? It tracks separately than callerid in the cdr. so in your sip.conf, add an accountcode= statement for each sip entry, and in the AccountCode field in the CDR, you'll have the correct entry needed to determine who made the call. -Chris Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? snip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
