RE: [asterisk-users] incoming SIP call
If your SIP server loses REGISTERs then it cant place an inbound SIP call. Try changing the REGISTER frequency to lower value. When you see incoming SIP call fail, you might want to check whether the REGISTERs are working. Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Wednesday, April 18, 2007 11:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming SIP call Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: 7263e88c20c9f3 mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df6 6 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asteris k sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3 d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all all ow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freep honie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0..0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... !DSPAM:462643f450705772331342! ___ --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] incoming SIP call
Well, for outbound calls, the SIP Server challenges the INVITE with 401/407. Then Re-INVITE is sent which explains why outgoing call works. It is possible that the SIP Server doesnt check to see whether the caller is Registered. For inbound call, the SIP server needs to know the gateway contact information, and it is obtained only through REGISTER (if not statically configured in the SIP server, which is very unlikely). Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Thursday, April 19, 2007 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] incoming SIP call Hello and thanks for answering, As I just answer to Yuan LIU, what I don't understand, is that I can place an outbound call from asterisk to a gsm at the same time I can't get asterisk thought a inbound call. But I'll try what you advice me. I'll tell you the result of it Jean-Marc LE FEVRE Le 19 avr. 07 à 20:08, Bala Neelakantan a écrit : If your SIP server loses REGISTERs then it cant place an inbound SIP call. Try changing the REGI STER frequency to lower value. When you see incoming SIP call fail, you might want to check whether the REGISTERs are working. Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Wednesday, April 18, 2007 11:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming SIP call Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: lt;sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- S IP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: 7263e88c20c9f3 mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc 208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df6 6 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asteris k sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3 d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all all ow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband quali fy=6 fromdomain=freephonie.net [freep honie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all /P deny=0.0.0.0/0..0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion
RE: [asterisk-users] Huh? IP address ending with 611
Looks like a PolycomSoundPointIP bug to me. The Via header, Contact both has 66.38.177.611:5060 Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Sent: Friday, April 13, 2007 7:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Huh? IP address ending with 611 The weird thing is that the phone actually works for now, but I want to proactively fix anything that may go wrong (this phone _has_ to work until Saturday) A SIP debug gives me this: --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms hd-t3143cl*CLI sip -- SIP read from 66.38.177.61:5060: REGISTER sip:pbx.test.ca SIP/2.0 Via: SIP/2.0/UDP 66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701 From: notarius-phone-1 sip:[EMAIL PROTECTED];tag=28A4D21C-80220BAB To: sip:[EMAIL PROTECTED] CSeq: 1803 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Authorization: Digest username=notarius-phone-1, realm=asterisk, nonce=20d11a72, uri=sip:pbx.test.ca, response=dbcfab79977a81ea3681bbe574bd1c37, algorithm=MD5 Max-Forwards: 70 Expires: 30 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Apr 13 08:28:26 WARNING[20348]: chan_sip.c:7036 check_via: '66.38.177.611' is not a valid host Transmitting (NAT) to 66.38.177.61:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701;received=66.38.177.61 From: notarius-phone-1 sip:[EMAIL PROTECTED];tag=28A4D21C-80220BAB To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Seq: 1803 REGISTERp User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 66.38.177.61:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701;received=66.38.177.61 From: notarius-phone-1 sip:[EMAIL PROTECTED];tag=28A4D21C-80220BAB To: sip:[EMAIL PROTECTED];tag=as42c283c1 Call-ID: [EMAIL PROTECTED] CSeq: 1803 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 30 Contact: sip:[EMAIL PROTECTED]:5060;expires=30 Date: Fri, 13 Apr 2007 12:28:26 GMT Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 23:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Huh? IP address ending with 611 Can you do a packet capture and see what the actual contact (Via) in fact says right before it hits Asterisk? On Thu, 12 Apr 2007, Mike said something to this effect: Hi, I`m getting this (from one of my registered phone that has been installed at some location I can`t access at the moment) in the Asterisk CLI. Notice the last 3 digits of the IP address in the error message: Apr 12 23:06:16 WARNING[25593]: chan_sip.c:7036 check_via: '67.39.117.611' is not a valid host Of course it's not a valid host! But, when using sip show peers, the phone is actually listed with IP address 67.39.117.61 (which makes alot more sense, but then again I shouldn`t be getting any warning in this case). Where is that error coming from then? Are there any consequences? Using 1.2.13. Mike -- Alex Balashov [EMAIL PROTECTED] ___ --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 ___ --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 ___ --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] Remote host can't match request NOTIFY to call
Looks like, the SIP NOTIFY message is getting a 481 Call leg does not exist response. You can ignore this message. But it will be interesting to see the full sip debug output to see what is going on. Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Richard Klingler Sent: Saturday, March 24, 2007 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Remote host can't match request NOTIFY to call Evnin'... Anybody got an idea where those CLI messages come from? [Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. Interestingly all are caused by local IP used by asterisk-1.4.1 cheers rick ___ --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 ___ --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 Registering to other SIP servers.
Hello, I am trying to REGISTER asterisk to a SIP server, which is listening on Port 6060 (not 5060). The sip.conf file contains register=1847420:[EMAIL PROTECTED]:6060/1847420 maxexpirey=3600 defaultexpirey=120 But the REGISTER message is sent to Port 6060, but the Request-URI still contains, 5060. This is being rejected by SIP server. REGISTER sip:192.168.2.94 SIP/2.0 The request line instead should be like this: REGISTER sip:192.168.2.94:6060 SIP/2.0 See the attached debug logs. SIP Debugging enabled *CLI reload Mar 7 13:23:12 NOTICE[17748]: cdr.c:1193 do_reload: CDR simple logging enabled. Mar 7 13:23:12 NOTICE[17748]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'us' Mar 7 13:23:12 WARNING[17748]: chan_zap.c:11029 setup_zap: Ignoring signalling Mar 7 13:23:12 WARNING[17748]: chan_zap.c:11029 setup_zap: Ignoring signalling *CLI Mar 7 13:23:12 NOTICE[17759]: chan_sip.c:5465 sip_reregister:-- Re-registration for [EMAIL PROTECTED] REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.2.94:6060: REGISTER sip:192.168.2.94 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.96:5060;branch=z9hG4bK1d8de05f;rport From: sip:[EMAIL PROTECTED];tag=as40ac92ca To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 Thanks, Neel ___ --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] chan_sip.c:10173 handle_response: Dont knowhowtohandle a 202 Accepted respons
Yuan,
It looks like you are getting 202 for SIP Request method MESSAGE. The 202
response is processed properly. Need to see the message fully. You can
capture sip debug if you don't have ethereal. This will provide more detail
call flow.
.{N/MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/U
13:42:12.761685 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 468
E:[EMAIL PROTECTED]
..
...SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.0.0.201:5060
13:42:12.793347 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 399
E;[EMAIL PROTECTED]
..
Thanks,
Neel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Tuesday, February 27, 2007 4:23 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] chan_sip.c:10173 handle_response: Dont
knowhowtohandle a 202 Accepted respons
From: Bala Neelakantan [EMAIL PROTECTED]
Date: Tue, 27 Feb 2007 14:21:32 -0600
Looks like asterisk is receiving 202 while it is not expecting it.
/*! \brief Handle SIP response in dialogue */
/* XXX only called by handle_request */
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct
sip_request *req, int ignore, int seqno)
Can you provide ethereal capture when you see this log message?
Neel,
Thanks for the reply. I don't have ethereal on the machine and not sure how
to capture - non-graphic terminal environment. Below is output from
tcpdump. In this session, I see two 202 Accepted from 1.4.0, only one
don't know notice. Interestingly, identical tests between two 1.2.13
Asterisk does not produce this.
I assume that this is nothing serious, because the session completes without
any problem, and the message is only a notice. If anything, I'll simply
revert to 1.2. (These are non-production.)
Yuan Liu
13:42:12.685850 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 749
E.. [EMAIL PROTECTED]
..
...INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1
13:42:12.686783 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 430
E...D[EMAIL PROTECTED] +
...
..
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.10:5060;br
13:42:12.687705 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 710
E...D'[EMAIL PROTECTED]
...
..
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.10:5060;branch
13:42:12.688229 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 363
[EMAIL PROTECTED]
..
s.WACK sip:[EMAIL PROTECTED] SIP/2.0
V
ia: SIP/2.0/UDP 10.0
13:42:12.761105 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 371
E...D([EMAIL PROTECTED] d
...
..
.{N/MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/U
13:42:12.761685 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 468
E:[EMAIL PROTECTED]
..
...SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.0.0.201:5060
13:42:12.793347 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 399
E;[EMAIL PROTECTED]
..
..{MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
V
ia: SIP/2.0/UDP
13:42:12.793863 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 448
E...D)[EMAIL PROTECTED] .
...
..
...oSIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.0.0.10:5060;
13:42:12.796133 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 332
[EMAIL PROTECTED] .
...
..
.Tw.BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1
13:42:12.796777 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 463
E[EMAIL PROTECTED]
..
...SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.201:5060;branc
Thanks,
Neel
-Original Message-
What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from
1.4.0.)
Yuan Liu
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RE: [asterisk-users] chan_sip.c:10173 handle_response: Dont know how tohandle a 202 Accepted respons
Looks like asterisk is receiving 202 while it is not expecting it. /*! \brief Handle SIP response in dialogue */ /* XXX only called by handle_request */ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno) Can you provide ethereal capture when you see this log message? Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Tuesday, February 27, 2007 1:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_sip.c:10173 handle_response: Dont know how tohandle a 202 Accepted respons What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu ___ --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 ___ --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] deprecated - CLI help vs. source code
Usage of LookupBlackList is deprecated. This means, the usage will work,
but there is no guarantee that it will work in future. You might want to
try using BLACKLIST() instead.
Thanks,
Neel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: Monday, February 26, 2007 3:07 PM
To: asterisk-dev@lists.digium.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] deprecated - CLI help vs. source code
Could someone with inside knowledge comment on that? If the
source code says deprecated but the CLI help does not mention
that - whom do I trust?
Original message
Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions
From: Philipp Kempgen [EMAIL PROTECTED]
Thomas Kenyon wrote:
Philipp Kempgen wrote:
You might use the Blacklist() application in 1.2 (deprecated!).
Using AstDB is an option:
Is it really deprecated? I use LookupBlacklist (tested as working) in
1.4, and show application LookupBlacklist doesn't mention it.
But the source code (app_lookupblacklist.c) has this line:
ast_log(LOG_WARNING, LookupBlacklist is deprecated. Please use
${BLACKLIST()} instead.\n);
So I'm not sure if it's deprecated or not. But this points us
to a new function: BLACKLIST()
[...]
Regards,
Philipp
--
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Let's use IT to solve problems and not to create new ones.
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