Hi All,
I am having this weird issue.
Outbound routes working fine,dialing using the correct trunk, When ringing the quality is perfect. But when the call is picked the call quality drops and I can’t hear any thing.
I am using pfingo as our SIP provider.
Here is my trunk settings.
host=sip.pfingo.com
username=XXXXXX
secret=XXXXXXX
type=peer
fromdomain=sip.pfingo.com
fromuser=XXXXX
nat=yes
please advise.
Thank you so much.
Depending on details of the SIP transaction, the ringback tone was probably generated by Asterisk or by your device, so it’s not surprising it sounds perfect.
Which is it? If you hear complete silence, that’s likely a blocked port or NAT issue. However, if you have choppy or distorted voice, that’s likely related to insufficient bandwidth, packet loss, codec issue, etc.
What does work? Do calls between extensions sound ok? Do you have other trunks? If so, do they sound ok? On problematic calls, can the remote party hear you ok? Have you tried other destinations with pfingo?
How is your Asterisk connected to the Internet? What kind of phone or other device are you calling from? How is it connected to Asterisk? If you configure the device to access pfingo directly, is the quality ok? What codec(s) are you trying to use?
Thank you Stewart1,
I can hear some thing , but it’s distorted. And I am having quite high bandwidth (100mbps).
My calls between extensions are perfect. I can call using dahdi lines ( analog) no issues. If i called the destination using pfingo (desktop) application. I can connect and talk with out any issues.
Asterisk is behind the firewall and I enabled all trafic from my network to go out to WAN with out any filters.
I am using ulaw,alaw,gsm codecs in asterisk.
Appreciate any help in the above question.
I’m very puzzled and would expect your quality to be excellent. Try turning on sip debug and see whether you can spot something strange in the SDP, e.g. wrong packetization or codec.
Otherwise, we can look at the incoming RTP and see what is wrong: Enable only ulaw, alaw. Also disable canreinvite. Then, use tcpdump to capture a failing call. Copy the file to your PC and open it in Wireshark. Select Telephony->VoIP Calls and click on your call. From there, you can examine and/or listen to the RTP.
Alternatively, you might just have Asterisk record the call. If the recorded audio is bad, you can look at it in detail with an audio editor and try to understand the problem. If it’s good, that should be a clue as to where the trouble lies. Or, get a SIP trunk from another provider, to see whether your problem is specific to pfingo. Many offer a small credit at signup, so you can test without making a payment, including Voxbeam, AnveoDirect, Flowroute, and Localphone.
On the bad calls, can the person you call hear you ok? Do incoming calls have the same quality problem? Are you in Singapore? What country(ies) are you calling? What kind(s) of IP phones or other devices are used for your extensions?