Hi,
i've just got 2 grandstream phones and when I try to connect them with *
I get the following:
-- Playing 'demo-abouttotry' (language 'en')
WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 02da8d3d-a8de-c7e1-5f07-c032ab23d909@10.1.1.179 for
seqno 59134 (Response)
I've seen there was some discussion on this already but i couldn't find
any resolution. Can anyone help?
Regards,
Robert Ivanc
Hi robert, I found that the disallow=all and then specify a codec with allow= was required in sip.conf. [17471234567] type=friend username=17471234567 secret=censored host=dynamic nat=yes disallow=all allow=ulaw Jon Hopper robert ivanc <Robert@netsec.si> wrote ..> Hi, > > i've just got 2 grandstream phones and when I try to connect them with > * > I get the following: > > -- Playing 'demo-abouttotry' (language 'en') > WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries > exceeded on call 02da8d3d-a8de-c7e1-5f07-c032ab23d909@10.1.1.179 for > seqno 59134 (Response) > > I've seen there was some discussion on this already but i couldn't find > any resolution. Can anyone help? > > Regards, > Robert Ivanc > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
What does this mean? I have a sipura 3000 with an analog line that I have created as a trunk. Incoming calls make it to the sipura but not to the pbx. However I can make outgoing calls but have no audio. I thought it might be a codec issue so I set disallow=<blank> and commented out the "allow=". I get the following in my logfile: build_route: Contact hop: satelliteout Jun 2 21:06:33 VERBOSE[2393]: -- SIP/satelliteout-af86 is ringing Jun 2 21:06:33 VERBOSE[2393]: -- SIP/satelliteout-af86 answered SIP/100-6eab Jun 2 21:06:33 VERBOSE[2393]: -- Attempting native bridge of SIP/100-6eab and SIP/satelliteout-af86 Jun 2 21:06:39 WARNING[2393]: Maximum retries exceeded on call c207e46a-baa6c1e4@192.168.5.100 for seqno 102 (Non-critical Response) Satelliteout is my outbound trunk and the call is being made from extension 100. Any idea what this means, I don't see anything that indicates an error when running asterisk -rvvvvv in the console, this is taken from the asterisk log files. Any help is greatly appreciated!
Periodically I will get this type of message in the * log: WARNING[18535]: Maximum retries exceeded on call 5dbbf3382b43aae51605b6cd3591c00a@1x.x.x.x for seqno 102 (Non-critical Response) The ip address listed sometimes is the * box itself and sometimes will be a sip cisco sip phone. When this happens often there will be lots of these messages, so 30 or 40 in 2 minutes time. I know it is a warning, but when this happens folks complain that calls are dropped or they can not hear. Asterisk suddenly lags in call completion and talk time. How can I go about finding what causes this? We have a normal call load of about 20 to 40 calls at a time, running 1.0.7 stable. We use the queue to handle incoming customer calls. And the AgentCallBackLogin. Doing a google revealed little that seemed helpful or I did not google right :) I have thought of removing the mpg123 stuff and using raw files. But, I don't know that that is the real problem. It seems especially strange that it complains about its on ip address sometimes. Any one else ever see this or have an idea where to trouble shoot it? -- respectfully, Joseph --------------------