sr-users August 2012

sr-users@lists.kamailio.org

77 participants
127 discussions

LCR - inconsistent return results for gateway selection.
by Graham Wooden (personal) 06 Aug '12

06 Aug '12

Hello, I am noticing that some of the time (not always consistent) that load_gw() / next_gw() will not always return the gateway with the lowest priority number. For example, when I look up a route for 205595, I have three grp_ids (gateways) that services the destination; with them being separated with 1, 2, and 3 priorities: id prefix from_uri grp_id priority 7468 205595 NULL 1 1 7469 205595 NULL 2 2 7470 205595 NULL 7 3 About 1/2 of … [View More]

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Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

Can you paste this output too? lsmod | grep -i sip -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Thanks for prompt reply. Here is the out put if command iptables -L -n: Chain INPUT (policy ACCEPT) … [View More]target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server. Vijay TH -============================================================================== Date: Mon, 06 Aug 2012 06:35:04 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To: sr-users(a)lists.sip-router.org Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com> Content-Type: text/plain; charset="utf-8" You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH [View Less]

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Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

Ah, you short-circuited where I was going with this. Couldn't remember the name of the module. Yep, Vijay. What Richard said. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Richard Fuchs <rfuchs(a)sipwise.com> wrote:On 08/06/12 06:08, Vijay Thakur … [View More]

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Kernel Droping SIP packet
by Vijay Thakur 06 Aug '12

06 Aug '12

Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=… [View More]

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Re: [SR-Users] B2BUA issues
by Alex Balashov 06 Aug '12

06 Aug '12

For your relatively narrow, specific applications in your topology, no. You'd be better off installing SEMS per se. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/phillman25 <phillman25(a)gmail.com> wrote:Hello Alex Will try with SEMS first, found something … [View More]called sip:provider CE v2.4 from http://www.sipwise.com/news/announcements/spce-v2_4-release/ if i'm not mistaken, this seems to combine Kamailio with SEMS? Do you think that this might be an easier installation rather than installing SEMS on its own as it seems to provide more documentation? Thanks again! On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ phillman25 <phillman25(a)gmail.com> wrote: Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip [View Less]

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Re: [SR-Users] Kernel Droping SIP packet
by Vijay Thakur 06 Aug '12

06 Aug '12

Thanks for prompt reply. Here is the out put if command iptables -L -n: *Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server. Vijay TH -============================================================================== * Date: Mon, … [View More]

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Re: [SR-Users] B2BUA issues
by Alex Balashov 06 Aug '12

06 Aug '12

The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E … [View More]Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/phillman25 <phillman25(a)gmail.com> wrote:Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip [View Less]

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Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-… [View More]

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Re: [SR-Users] B2BUA issues
by phillman25 06 Aug '12

06 Aug '12

Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for … [View More]these specific scenarios? >From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b. 66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx; lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip [View Less]

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B2BUA issues
by phillman25 06 Aug '12

06 Aug '12

Dear List I am trying to accomplish the following: Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) When trying the above scenario, the call is silent and drops after a few seconds. In syslog i observe the following error: *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing name-addr (sip:22030305@192.168.10.189:5060>)* … [View More]

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sr-users August 2012