Automated call completion testing?
Hi, folks, I'm on the hunt for an automated call completion testing solution. Criteria: Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter. Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold Must be able to test legacy phone numbers (e.g. +14154622991) Must be able to test via configurable routes Should be open-source Should be compatible with Asterisk Should be something I can integrate with one or more of the following: Zenoss, Nagios, AskoziaPBX, pfSense Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991). Any suggestions? thanks, Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com
I'd probably start with SIPsak (http://sipsak.org/) but it would be a tool in a script you wrote probably, not the end all be all. I'm not sure I've seen an open source solution that matches these requirements. In the paid world look at Spirent's Abacus. -Scott -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Graham Freeman Sent: Wednesday, June 09, 2010 4:28 PM To: voiceops at voiceops.org Subject: [VoiceOps] Automated call completion testing? Hi, folks, I'm on the hunt for an automated call completion testing solution. Criteria: Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter. Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold Must be able to test legacy phone numbers (e.g. +14154622991) Must be able to test via configurable routes Should be open-source Should be compatible with Asterisk Should be something I can integrate with one or more of the following: Zenoss, Nagios, AskoziaPBX, pfSense Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991). Any suggestions? thanks, Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
Ditto on sipsak and rolling your own. An additional tool comes from way back in the Age of the Mullet. The "internet edge" was mostly dialup modems back then. If you're willing to slap an ATA on one end, you can find a bunch of modem (war dialing, dialback, shared pool, etc) scripts out there. David On Wed, Jun 9, 2010 at 1:55 PM, Scott Berkman <scott at sberkman.net> wrote:
I'd probably start with SIPsak (http://sipsak.org/) but it would be a tool in a script you wrote probably, not the end all be all. ?I'm not sure I've seen an open source solution that matches these requirements. ?In the paid world look at Spirent's Abacus.
? ? ? ?-Scott
-----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Graham Freeman Sent: Wednesday, June 09, 2010 4:28 PM To: voiceops at voiceops.org Subject: [VoiceOps] Automated call completion testing?
Hi, folks,
I'm on the hunt for an automated call completion testing solution.
Criteria:
Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter.
Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold
Must be able to test legacy phone numbers (e.g. +14154622991)
Must be able to test via configurable routes
Should be open-source
Should be compatible with Asterisk
Should be something I can integrate with one or more of the following: Zenoss, Nagios, AskoziaPBX, pfSense
Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991).
Any suggestions?
thanks,
Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
I would look at the CDR or Radius accounting message . Whats wrong in relying in CDR/radius info? -N ________________________________ From: Graham Freeman <graham.freeman at cernio.com> To: voiceops at voiceops.org Sent: Wed, June 9, 2010 4:27:37 PM Subject: [VoiceOps] Automated call completion testing? Hi, folks, I'm on the hunt for an automated call completion testing solution. Criteria: Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter. Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold Must be able to test legacy phone numbers (e.g. +14154622991) Must be able to test via configurable routes Should be open-source Should be compatible with Asterisk Should be something I can integrate with one or more of the following: Zenoss, Nagios, AskoziaPBX, pfSense Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991). Any suggestions? thanks, Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
Perhaps "recqual" is what you're looking for. End-to-end testing with tones, comparing the recorded tone with the transmitted tone. Works with Asterisk. Implies you can control both "end" of the call, since otherwise it is extremely difficult to conclude that a call is "completed" since many errors are not obvious if all you're looking for is a media path starting up. http://www.voip-info.org/wiki/view/Recqual JT On Jun 9, 2010, at 1:27 PM, Graham Freeman wrote:
Hi, folks,
I'm on the hunt for an automated call completion testing solution.
Criteria:
Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter.
Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold
Must be able to test legacy phone numbers (e.g. +14154622991)
Must be able to test via configurable routes
Should be open-source
Should be compatible with Asterisk
Should be something I can integrate with one or more of the following: Zenoss, Nagios, AskoziaPBX, pfSense
Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991).
Any suggestions?
thanks,
Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
I went with sipp for that exact same purpose. I had to script a bit around it to integrate with my zabbix monitoring but now I can get alerts on failed calls. On Wed, Jun 9, 2010 at 5:48 PM, John Todd <jtodd at loligo.com> wrote:
Perhaps "recqual" is what you're looking for. ?End-to-end testing with tones, comparing the recorded tone with the transmitted tone. ?Works with Asterisk. ? Implies you can control both "end" of the call, since otherwise it is extremely difficult to conclude that a call is "completed" since many errors are not obvious if all you're looking for is a media path starting up.
http://www.voip-info.org/wiki/view/Recqual
JT
On Jun 9, 2010, at 1:27 PM, Graham Freeman wrote:
Hi, folks,
I'm on the hunt for an automated call completion testing solution.
Criteria:
Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter.
Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold
Must be able to test legacy phone numbers (e.g. +14154622991)
Must be able to test via configurable routes
Should be open-source
Should be compatible with Asterisk
Should be something I can integrate with one or more of the following: ?Zenoss, Nagios, AskoziaPBX, pfSense
Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991).
Any suggestions?
thanks,
Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
One of the most bizzare things about sipp is its placement of XML configurations into string constants in the code. On Jun 11, 2010, at 12:27 AM, Antoine Reversat <a.reversat at gmail.com> wrote:
I went with sipp for that exact same purpose. I had to script a bit around it to integrate with my zabbix monitoring but now I can get alerts on failed calls.
On Wed, Jun 9, 2010 at 5:48 PM, John Todd <jtodd at loligo.com> wrote:
Perhaps "recqual" is what you're looking for. End-to-end testing with tones, comparing the recorded tone with the transmitted tone. Works with Asterisk. Implies you can control both "end" of the call, since otherwise it is extremely difficult to conclude that a call is "completed" since many errors are not obvious if all you're looking for is a media path starting up.
http://www.voip-info.org/wiki/view/Recqual
JT
On Jun 9, 2010, at 1:27 PM, Graham Freeman wrote:
Hi, folks,
I'm on the hunt for an automated call completion testing solution.
Criteria:
Must test call completion, not just whether the SIP gateway IP is reachable and has low ping/jitter.
Must be able to notify defined points of contact (ideally via email) upon reaching a set failure threshold
Must be able to test legacy phone numbers (e.g. +14154622991)
Must be able to test via configurable routes
Should be open-source
Should be compatible with Asterisk
Should be something I can integrate with one or more of the following: Zenoss, Nagios, AskoziaPBX, pfSense
Should be able to test both SIP URIs (e.g. sip:firstname.lastname at cernio.com) and legacy phone numbers (e.g. +14154622991).
Any suggestions?
thanks,
Graham Freeman Cernio Technology Cooperative www.cernio.com graham.freeman at cernio.com
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
participants (7)
-
a.reversat＠gmail.com -
abalashov＠evaristesys.com -
graham.freeman＠cernio.com -
hiersd＠gmail.com -
jtodd＠loligo.com -
naushit＠yahoo.com -
scott＠sberkman.net