Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: Teviot on November 23, 2008, 11:54:08 AM
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Well I'm stuck. I have installed SAIL SME with no problem and can dial my UK base sip provider's echo test and hear that fine. I can't seem to get my voice to be played back to me after speeking when prompted. Also, I can't talk between extension locally. Dialing is not an issue.
The log entries are as follows
[Nov 24 15:25:01] NOTICE[4701] chan_sip.c: Peer '30134831' is now Reachable. (376ms / 2000ms)
[Nov 24 15:25:43] WARNING[4701] chan_sip.c: Maximum retries exceeded on transmission d430d577-c7a6cbae@192.168.76.43 for seqno 102 (Critical Response)
[Nov 24 15:25:43] WARNING[4701] chan_sip.c: Hanging up call d430d577-c7a6cbae@192.168.76.43 - no reply to our critical packet.
I don't know why but I'm thinking that this might be a codec issue
Please help
Teviot
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Hi
Just tried a complete re-install and had the system talking both directions to my VOIP provider in the UK. The added another trunk for an Australian VOIP account I have and the system stopped transmitting audio.
I really need help with this
Teviot
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usually one-way sound is a reult of the localnet parameter being incorrectly set in the sip.conf header.
The default is 192.168.1.0
Also, if you are running server-only, then make sure you have the correct external ip address filled out in Globals
Usually one or both those two.
Best
S
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Thank you selintra
I fixed that problem and audio is now flowing
Teviot
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usually one-way sound is a result of the localnet parameter being incorrectly set in the sip.conf header.
The default is 192.168.1.0
Ahhh... That fixed the problem I had with some Siemens Gigaset S685IP phones. They worked fine when connected directly to my internet Router (and registering with the SME box as remote phones) but not when inside my SME box (which is in server and gateway mode). Spent several evenings tearing my hair out till I spotted this answer. :???:
Most frustrating because we have an almost identical system at work and it works fine, that system was supplied by Selintra and whilst their support has always been first class I did not feel it was reasonable to call them up for tech support on my home brew system :P
Mark Leman
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Selintra, IMHO this setting in the sip.conf header should really be auto-adjusted by SAIL to match whatever the local settings are. SAIL is supposed to automate as much as possible, and this is something that deserves automation.