Hello: I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 106 to 107 I get the followng error :
1. in the asterisk console: " Unresolvable destination (478/SL)" 2. in the kamailio log: May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]: failed to resolve "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL)
Any idea about the cause of this problem?
ps:kamailio and asterisk are running in the same machine.
ps:attachment is Asterisk CLI log
Best Regards, zhengyw
郑友闻 医疗终端部 东软熙康健康有限公司 沈阳市浑南新区新秀街2号，A1楼216 Postcode：110179 Tel：+86 24 83662278 Mobile：15242493836 Email : zhengyw@neusoft.com http://www.neusoft.com --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------
Hello
On 5/8/13 11:20 AM, zhengyw wrote:
Hello: I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 106 to 107 I get the followng error :
- in the asterisk console: " *Unresolvable destination (478/SL)"
*2. in the kamailio log:
- May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR:
<core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]: failed to resolve "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL)* Any idea about the cause of this problem?
the invite comes with a bad domain in R_URI, see your logs:
Reliably Transmitting (no NAT) to 10.11.2.47:5060: INVITEsip:106@(null) SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4 Route:sip:10.11.2.47;lr=on;ftag=29997
Probably something is not properly configured in asterisk.
Cheers, Daniel
hello daniel: thank you very much! but I can't find the problem in the asterisk. attachment is asterisk's configure file, kamailio's configure file and data.can you help with this problem?
ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04
Best Regards, zhengyw kamailio.cfg http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg sip.conf http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf extconfig.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf extensions.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf db_result.txt http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt
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Hello,
I am not that familiar to troubleshoot asterisk configuration files, but from logs I could see the resulting URI is:
INVITEsip:106@(null) SIP/2.0
That is wrong, meaning something incorrect is done when setting it in asterisk. Maybe someone else can help more with asterisk.
Cheers, Daniel
On 5/13/13 4:02 AM, zhengyw wrote:
hello daniel: thank you very much! but I can't find the problem in the asterisk. attachment is asterisk's configure file, kamailio's configure file and data.can you help with this problem?
ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04Best Regards, zhengyw kamailio.cfg http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg sip.conf http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf extconfig.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf extensions.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf db_result.txt http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt
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Hello:
I have followed the steps in the guide(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb):
1. Could you find any mistake in my configuration file?
2. I also wonder how could I be sure of "Be sure you update the listen IP and port as well if Asterisk is running on the same
system with Kamailio" and "Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio".
thank you very much
PS：attachment is kamailio、asterisk's congifure file.
Best Regards
zhengyw
----- Original Message ----- From: "Daniel-Constantin Mierla" miconda@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Monday, May 13, 2013 3:26 PM Subject: Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
Hello,
I am not that familiar to troubleshoot asterisk configuration files, but from logs I could see the resulting URI is:
INVITEsip:106@(null) SIP/2.0
That is wrong, meaning something incorrect is done when setting it in asterisk. Maybe someone else can help more with asterisk.
Cheers, Daniel
On 5/13/13 4:02 AM, zhengyw wrote:
hello daniel: thank you very much! but I can't find the problem in the asterisk. attachment is asterisk's configure file, kamailio's configure file and data.can you help with this problem?
ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04Best Regards, zhengyw kamailio.cfg http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg sip.conf http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf extconfig.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf extensions.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf db_result.txt http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt
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On 13 May 2013 09:46, zhengyw zhengyw@neusoft.com wrote:
Hello:
I have followed the steps in the guide(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb ):
1. Could you find any mistake in my configuration file?
I don't see any obvious errors apart form the incorrect table structure in video1_sipregs which I already mentioned.
2. I also wonder how could I be sure of "Be sure you update thelisten IP and port as well if Asterisk is running on the same
You do this using the bindaddr/udpbindaddr/tcpbindaddr settings in sip.conf. You have "udpbindaddr=10.11.2.47:5080" set. Perhaps it would help if you instead simply set "bindaddr=10.11.2.47:5080" - it will have the same affect, as tcp is disabled anyway.
system with Kamailio" and "Be sure you configure Asterisk to notauthenticate SIP requests coming from Kamailio".
This is achieved by having no entry in the "secret" field of the video1_sipusers table, so that Asterisk does not try to authenticate the peer, as Kamailio has already done so.
-Barry
thank you very much PS：attachment is kamailio、asterisk's congifure file.Best Regards
zhengyw
----- Original Message ----- From: "Daniel-Constantin Mierla" miconda@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Monday, May 13, 2013 3:26 PM Subject: Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
Hello,
I am not that familiar to troubleshoot asterisk configuration files, but from logs I could see the resulting URI is:
INVITEsip:106@(null) SIP/2.0
That is wrong, meaning something incorrect is done when setting it in asterisk. Maybe someone else can help more with asterisk.
Cheers, Daniel
On 5/13/13 4:02 AM, zhengyw wrote:
hello daniel: thank you very much! but I can't find the problem in the asterisk. attachment is asterisk's configure file, kamailio's configure file
and
data.can you help with this problem?
ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version12.04
Best Regards, zhengyw kamailio.cfg http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg sip.conf <http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf
extconfig.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf extensions.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf db_result.txt http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt
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thank you very much. but I don't know What is the cause of the problem*（" Unresolvable destination (478/SL)" *）?
1.I was very confused, don't know what to do. can you give more hints ？
2.When have you done that *Asterisk and Kamailio Realtime Integration using Asterisk Database*?
zhengyw
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On 16 May 2013 02:18, zhengyw zhengyw@neusoft.com wrote:
thank you very much. but I don't know What is the cause of the problem*（" Unresolvable destination (478/SL)" *）?
1.I was very confused, don't know what to do. can you give more hints ？
Not really. You have not confirmed that you have done what I suggested already, nor provided the log files I suggested.
2.When have you done that *Asterisk and Kamailio Realtime Integration using Asterisk Database*?
I have integrated Kamailio and Asterisk Realtime many times, although I have not followed the (very good) guide that you did, nor have I ever done it with both Asterisk and Kamailio on the same server.
-Barry
zhengyw
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I think - best integration, is - Kamailio handles users, Asterisk handles what it must handle - like IVR, registration with VoIP providers etc.... At least we do things this way - Asterisk dont know nothing about users. All calls from Kamailio are accepted, and forwarded either back to Kamailio, if call is to other user, or processed properly. If second user is not online - call drops, and Asterisk understand that second user is not registered. This of course is not best solution, but it works fine for me.
On Thu, May 16, 2013 at 12:44 PM, Barry Flanagan barry@flanagan.ie wrote:
On 16 May 2013 02:18, zhengyw zhengyw@neusoft.com wrote:
thank you very much. but I don't know What is the cause of the problem*（" Unresolvable destination (478/SL)" *）?
1.I was very confused, don't know what to do. can you give more hints ？
Not really. You have not confirmed that you have done what I suggested already, nor provided the log files I suggested.
2.When have you done that *Asterisk and Kamailio Realtime Integration using Asterisk Database*?
I have integrated Kamailio and Asterisk Realtime many times, although I have not followed the (very good) guide that you did, nor have I ever done it with both Asterisk and Kamailio on the same server.
-Barry
zhengyw
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Hi Barry: This issue have not been resolved after following by your method that modifed the video1_sipregs table struct,attachment is table info and asterisk log. Can you help me with this problem? thank you very much!
best zhengyw *asterisk log* asterisk_log.txt http://sip-router.1086192.n5.nabble.com/file/n118567/asterisk_log.txt *dbinfo.txt* dbinfo.txt http://sip-router.1086192.n5.nabble.com/file/n118567/dbinfo.txt
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On 20 May 2013 04:07, zhengyw zhengyw@neusoft.com wrote:
Hi Barry: This issue have not been resolved after following by your method that modifed the video1_sipregs table struct,attachment is table info and asterisk log. Can you help me with this problem? thank you very much!
I can see that your sipregs table is now being properly populated. I have no idea though why Kamailio is sending the Contact: as 106@(null), which appears to be the main issue.
<--- SIP read from UDP:10.11.2.47:5060 ---> INVITE sip:107@10.11.2.47 SIP/2.0 Record-Route: sip:10.11.2.47;lr=on;ftag=617 Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK7c6b.86d570e3.0 Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK2720 From: sip:106@10.11.2.47;tag=617 To: "107" sip:107@10.11.2.47 Call-ID: 5246 CSeq: 21 INVITE Contact: sip:106@(null) Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 69 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 349
You need to look in the Kamailio logs and try to see what the issue is on the initial INVITE
-Barry
best
zhengyw *asterisk log* asterisk_log.txt http://sip-router.1086192.n5.nabble.com/file/n118567/asterisk_log.txt *dbinfo.txt* dbinfo.txt http://sip-router.1086192.n5.nabble.com/file/n118567/dbinfo.txt
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Hi: The question has been settled! Thank you very much Barry and Gertjan!
best regard zhengyw
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On 27 May 2013 03:36, zhengyw zhengyw@neusoft.com wrote:
Hi: The question has been settled! Thank you very much Barry and Gertjan!
Glad to hear it.
For everyone's benefit could you please post what the actual issue was and how it was solved?
Thanks
best regard zhengyw
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Hi: Sorry. I reply late, because for I was unwell. I modified in 3 places, follow Barry and Gertjan:
1.modified video_sipregs table structor to correct. 2.update video_sipusers table's field[*context*] 'from sip' to *'NULL'*. 3.modified extensions.conf file. put the "*exten => _XXX,1,Dial(SIP/${EXTEN},60)"* under the [*default*].
best regard zhengyw
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On 13 May 2013 03:02, zhengyw zhengyw@neusoft.com wrote:
hello daniel: thank you very much! but I can't find the problem in the asterisk. attachment is asterisk's configure file, kamailio's configure file and data.can you help with this problem?
Hi
Your video1_sipregs table does not seem correct. Your registered users should have ipaddr and port fields populated. I notice you are missing a few fields which could be a problem.
mysql> select * from video1_sipregs; +----+------+-------------------------+--------+------+----------+-----------+------------+ | id | name | fullcontact | ipaddr | port | username | regserver | regseconds | +----+------+-------------------------+--------+------+----------+-----------+------------+ | 4 | 106 | sip:106@10.11.2.47:5060 | | 0 | 106 | | 1368169282 | | 5 | 107 | sip:107@10.11.2.47:5060 | | 0 | 107 | | 1368176017 | | 6 | 108 | | | 0 | | NULL | 0 | +----+------+-------------------------+--------+------+----------+-----------+------------+
The sipregs table should have this structure for Asterisk 10.7 as far as I know.
CREATE TABLE `sipregs` ( `id` INT(11) NOT NULL AUTO_INCREMENT, `name` VARCHAR(80) NOT NULL DEFAULT '', `fullcontact` VARCHAR(80) NOT NULL DEFAULT '', `ipaddr` VARCHAR(45) DEFAULT NULL, `port` mediumint(5) UNSIGNED NOT NULL DEFAULT '0', `username` VARCHAR(80) NOT NULL DEFAULT '', `regserver` VARCHAR(100) DEFAULT NULL, `regseconds` INT(11) NOT NULL DEFAULT '0', `defaultuser` VARCHAR(80) NOT NULL DEFAULT '', `useragent` VARCHAR(20) DEFAULT NULL, `lastms` INT(11) DEFAULT NULL, PRIMARY KEY (`id`), UNIQUE KEY `name` (`name`) );
-Barry
ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04
Best Regards, zhengyw kamailio.cfg http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg sip.conf http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf extconfig.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf extensions.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf db_result.txt http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt
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Barry Flanagan -
Daniel-Constantin Mierla -
Stoyan Mihaylov -
zhengyw