Run tcpdump with portrange 10000-20000 (or what range of ports you use for rtp) and dial test ivr to see what happening.
2011/10/17 gincantalupo <gincantal...@fgasoftware.com> > ** > Hi John, > > there is no firewall: > > snom <--> pbx <--> patton <--> pstn > > It happens ONLY with IVRs. Normal calls are fine. How can it be? > > I call my pbx from the customer pbx: when I directly call my phone it > works, when I call a test ivr it does not work...can a timing problem be the > cause??? > > Giorgio > > > > On 10/14/2011 03:48 PM, John Knight wrote: > > Hi Giorgio, > > This behavior usually indicates some sort of firewall issue where either > inbound or outbound rtp traffic (the voice) is being blocked or not routed > correctly, though the SIP traffic makes it through (as the call is being set > up correctly). This could also be multiple SIP extensions attempting to > register over the same port from a single location. > > What kind of firewall/router is being used at the location where these > Snoms are registering from? Are all the phones attempting to register over > port 5060 or are you setting them up to register over unique ports to > Asterisk? If you are setting them up to register over specific ports, are > they registering over those ports according to 'asterisk show peers'? Also, > is your asterisk box local or hosted somewhere? > > Comparing IAX2 to SIP registrations is somewhat different: IAX2 tends to > handle cutting through firewalls better though SIP is far better supported > by everyone. > > > > > On Fri, Oct 14, 2011 at 6:21 AM, gincantalupo < > gincantal...@fgasoftware.com> wrote: > >> Hi all, >> >> I'm stuck on a tricky problem. >> I set up an Asterisk 1.4.26.2 on a box with a bunch of Snom Phones. When I >> call an IVR I get the damned one way voice phenomena. It is not randomic, it >> happens all the time. >> I tried to upgrade the snom firmware to 7.3.30 but nothing changed. >> If I call a phone I get a normal conversation and no problem occurs if I >> (blind) transfer the call. >> If I use a IAX phone everything is fine. >> I think it is a SIP problem but I checked the sip files and they seem ok. >> Tones seems to pass since the caller (me) can make a choice from within >> the IVR menu. >> >> Sincerely, I haven't any idea left to try... >> >> Any hints? >> >> Thanks >> >> Giorgio >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- > * > * > > *John Knight* > Classic City Telco LLC > *Email:* j...@classiccitytelco.com | *Main:* (706) 995-0200 > *Direct:* (706) 995-0201 | *Mobile:* (706) 255-9203 > > * * > > > -- > _____________________________________________________________________ > -- 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 >
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-- _____________________________________________________________________ -- 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