Re: [asterisk-users] how to check Asterisk SIP registration
You've unfortunately gotten a lot of confused answers. To try to clear this up: 1. Only type=peer objects accept registrations. sip show users or sip show registry has nothing to do with peers. A peer might be part of a type=friend 2. If you see IP addresses when you run sip show peers then those objects have an active registration, Asterisk knows where to reach them. 3. Nat's or firewalls between the device and Asterisk might cause issues with Asterisk sending messages to them or devices sending messages to Asterisk 4. Your output below indicates that Asterisk doesn't know how to reach the device, that Asterisk has no IP and port address to send messages to, thus the device is not registered at all. 5. Turning qualify on can help with keeping a NAT binding open. To summarize, start with looking for IP address in sip show peers. If we have an IP address, check the result of the Qualify option in the same output. If there's an IP, the device could reach Asterisk. If the status is unreachable Asterisk could not reach the device on the IP address. Then go hunting in your network to find the issue. Best regards, /Olle 24 dec 2009 kl. 17.39 skrev Vieri: Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- 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 --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- 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
Re: [asterisk-users] how to check Asterisk SIP registration
I appreciate everyone's feedback. I did not post the sip show peers output because I did not have time to save it but I'm fairly sure that qualify was OK and that IP addresses did show up. NAT/firewall is not an issue because Asterisk and the sip devices are on the same network (open LAN). Anyway, regardless of the sip show peers output, the fact that the SIP devices registered fine and communication was re-established after killing asterisk and starting it, demonstrates that the root cause is not the network but the Asterisk's SIP service. I am using an alias IP address on the SIP server. Usually it works fine but maybe this time something went wrong. At the time I had my issue, I checked that the alias IP address was defined. Maybe Asterisk's SIP service was not correctly bound/listening to that alias IP address... Maybe removing and adding the alias IP address would have magically solved the issue but I did not try that. Can the SIP service be restarted without affecting the rest of Asterisk? (I don't think sip reload does this) Thanks, Vieri --- On Sat, 12/26/09, Olle E. Johansson o...@edvina.net wrote: You've unfortunately gotten a lot of confused answers. To try to clear this up: 1. Only type=peer objects accept registrations. sip show users or sip show registry has nothing to do with peers. A peer might be part of a type=friend 2. If you see IP addresses when you run sip show peers then those objects have an active registration, Asterisk knows where to reach them. 3. Nat's or firewalls between the device and Asterisk might cause issues with Asterisk sending messages to them or devices sending messages to Asterisk 4. Your output below indicates that Asterisk doesn't know how to reach the device, that Asterisk has no IP and port address to send messages to, thus the device is not registered at all. 5. Turning qualify on can help with keeping a NAT binding open. To summarize, start with looking for IP address in sip show peers. If we have an IP address, check the result of the Qualify option in the same output. If there's an IP, the device could reach Asterisk. If the status is unreachable Asterisk could not reach the device on the IP address. Then go hunting in your network to find the issue. Best regards, /Olle 24 dec 2009 kl. 17.39 skrev Vieri: Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43
Re: [asterisk-users] how to check Asterisk SIP registration
Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- 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
Re: [asterisk-users] how to check Asterisk SIP registration
sip show registry might be more helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to check Asterisk SIP registration Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- 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 ___ -- 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
Re: [asterisk-users] how to check Asterisk SIP registration
Thanks but sip show registry yields nothing. --- On Thu, 12/24/09, Danny Nicholas da...@debsinc.com wrote: sip show registry might be more helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to check Asterisk SIP registration Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] how to check Asterisk SIP registration
I wrote a script to check clients and restart asterisk if registrations died (external IAX)...but you could modify for your needs. Check it out on www.generationd.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 12:06 PM To: Asterisk Users List Subject: Re: [asterisk-users] how to check Asterisk SIP registration Thanks but sip show registry yields nothing. --- On Thu, 12/24/09, Danny Nicholas da...@debsinc.com wrote: sip show registry might be more helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to check Asterisk SIP registration Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] how to check Asterisk SIP registration
Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- 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 ___ -- 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
