Re: [asterisk-users] SIP bad request
Just a follow-up in case somebody else sees this: I upgraded the Polycom phone to the latest firmware, that did it. I had been on the same version for almost a year without problems, so I don`t know if it`s the firmware version that was the issue or simply formatting the phone to factory default would have fixed it . Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, April 29, 2011 11:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP bad request What I am looking for? Here is a snippet, with some info obfuscated. I can see the bad request, but why there is such a message isn’t obvious. --- SIP read from UDP:23.23.23.23:23725 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: sip:user@192.168.1.90:5060 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Allow-Events: talk,hold,conference Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 - --- (11 headers 0 lines) --- --- SIP read from UDP:23.23.23.23:23725 --- SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: sip:user@192.168.1.90:5060 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ??? ? Sent: Friday, April 29, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP bad request Try to look in 'sip set debug peer user'. On 29.04.2011 18:10, Mike wrote: Hi, I have been getting reports phones ringing only a tiny moment and then going to voicemail. CLI output shows: -- SIP/user-0006fcdd is ringing -- Got SIP response 400 Bad Request back from 23.23.23.23 -- SIP/user-0006fcdd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which does explain it. How can I find the root cause of “bad request”? Call-limit is very high for this sip user, so I`m not reaching that limit for sure. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP bad request
may be the ip phone has the problem, try reset as factory On Fri, Apr 29, 2011 at 8:03 PM, Mike l...@net-wall.com wrote: What I am looking for? Here is a snippet, with some info obfuscated. I can see the bad request, but why there is such a message isn’t obvious. --- SIP read from UDP:23.23.23.23:23725 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: sip:user@192.168.1.90:5060 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Allow-Events: talk,hold,conference Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 - --- (11 headers 0 lines) --- --- SIP read from UDP:23.23.23.23:23725 --- SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: sip:user@192.168.1.90:5060 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *??? ? *Sent:* Friday, April 29, 2011 10:49 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP bad request Try to look in 'sip set debug peer user'. On 29.04.2011 18:10, Mike wrote: Hi, I have been getting reports phones ringing only a tiny moment and then going to voicemail. CLI output shows: -- SIP/user-0006fcdd is ringing -- Got SIP response 400 Bad Request back from 23.23.23.23 -- SIP/user-0006fcdd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which does explain it. How can I find the root cause of “bad request”? Call-limit is very high for this sip user, so I`m not reaching that limit for sure. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP bad request
Hi, I have been getting reports phones ringing only a tiny moment and then going to voicemail. CLI output shows: -- SIP/user-0006fcdd is ringing -- Got SIP response 400 Bad Request back from 23.23.23.23 -- SIP/user-0006fcdd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which does explain it. How can I find the root cause of bad request? Call-limit is very high for this sip user, so I`m not reaching that limit for sure. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP bad request
Try to look in 'sip set debug peer user'. On 29.04.2011 18:10, Mike wrote: Hi, I have been getting reports phones ringing only a tiny moment and then going to voicemail. CLI output shows: -- SIP/user-0006fcdd is ringing -- Got SIP response 400 Bad Request back from 23.23.23.23 -- SIP/user-0006fcdd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which does explain it. How can I find the root cause of “bad request”? Call-limit is very high for this sip user, so I`m not reaching that limit for sure. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP bad request
What I am looking for? Here is a snippet, with some info obfuscated. I can see the bad request, but why there is such a message isn’t obvious. --- SIP read from UDP:23.23.23.23:23725 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: sip:user@192.168.1.90:5060 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Allow-Events: talk,hold,conference Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 - --- (11 headers 0 lines) --- --- SIP read from UDP:23.23.23.23:23725 --- SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: sip:user@192.168.1.90:5060 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ??? ? Sent: Friday, April 29, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP bad request Try to look in 'sip set debug peer user'. On 29.04.2011 18:10, Mike wrote: Hi, I have been getting reports phones ringing only a tiny moment and then going to voicemail. CLI output shows: -- SIP/user-0006fcdd is ringing -- Got SIP response 400 Bad Request back from 23.23.23.23 -- SIP/user-0006fcdd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which does explain it. How can I find the root cause of “bad request”? Call-limit is very high for this sip user, so I`m not reaching that limit for sure. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Bad request protocol Packet on Asterisk 1.4.18
Remco Barendse wrote:
Hi all!!
I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk
1.4.18. Both are home PBX's and both boxes register to a SIP DID at
exactly same provider. One box runs without errors on the console, the
other box keeps repeating :
[Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705
determine_firstline_parts: Bad request protocol Packet
When i set debug on, it seems to come from that SIP DID.
--- SIP read from 82.101.62.99:5060 ---
Cirpack KeepAlive Packet
-
[Feb 11 23:37:59] WARNING[11292]: chan_sip.c:6705
determine_firstline_parts: Bad request protocol Packet
--- (1 headers 0 lines) ---
What i don't understand is why i get this message on one box only?
Ideas anyone?
Not sure. Difference in core set verbose? Afaik it's harmless but
quite annoying. Attached are two patches. One should fix the keep
alive stuff by silently dropping it and the other is a dtmf fix cause
Cirpack has a bug with dtmf. At least it did a while ago. Not sure if
it's still needed.
Regards,
Patrick
diff -uNr asterisk-1.4.13.org/channels/chan_sip.c asterisk-1.4.13/channels/chan_sip.c
--- asterisk-1.4.13.org/channels/chan_sip.c 2007-10-10 16:42:00.0 +0200
+++ asterisk-1.4.13/channels/chan_sip.c 2007-11-14 04:33:05.0 +0100
@@ -6620,6 +6620,12 @@
if (*e)
*e++ = '\0';
e = ast_skip_blanks(e);
+ if (!strcasecmp(req-rlPart1, Cirpack)
+ !strcasecmp(req-rlPart2, KeepAlive)
+ !strcasecmp(e, Packet)) {
+ /* Silently drop bogus Cirpack keepalive packets */
+return -1;
+ }
if (strcasecmp(e, SIP/2.0) ) {
ast_log(LOG_WARNING, Bad request protocol %s\n, e);
return -1;
diff -uNr asterisk-1.4.13.org/main/rtp.c asterisk-1.4.13/main/rtp.c
--- asterisk-1.4.13.org/main/rtp.c 2007-10-08 22:06:33.0 +0200
+++ asterisk-1.4.13/main/rtp.c 2007-11-11 13:12:28.0 +0100
@@ -1383,6 +1383,7 @@
[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
+ [96] = {0, AST_RTP_DTMF},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},
___
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[asterisk-users] SIP Bad request protocol Packet on Asterisk 1.4.18
Hi all!! I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk 1.4.18. Both are home PBX's and both boxes register to a SIP DID at exactly same provider. One box runs without errors on the console, the other box keeps repeating : [Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705 determine_firstline_parts: Bad request protocol Packet When i set debug on, it seems to come from that SIP DID. --- SIP read from 82.101.62.99:5060 --- Cirpack KeepAlive Packet - [Feb 11 23:37:59] WARNING[11292]: chan_sip.c:6705 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- What i don't understand is why i get this message on one box only? Ideas anyone? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
