Guideline for SIP Trunk Setup

Transcription

1 Guideline for SIP Trunk Setup with ZONETEL

2 Table of contents Sample sip.conf (it applies to asterisk 1.4.x)...3 Sample elastix setup... 3 Ports required... 4 Caller ID...4 FAQ... 5 After i dial out, the destination phone rings, but we could not hear each other... 5 How to dial to HK?... 5 How to make international call?...5 Sometimes, I could not receive incoming calls...6 I do not get a dial tone from zonetel...6 I can dial to HK, but not other destinations...6 The voice quality is not good... 6 I can dial out directly from phone, but call forward fails...7 Still have problem?...7

3 Sample sip.conf (it applies to asterisk 1.4.x) [general] Defaultexpiry=600 Register => [zonetel] Username=5804xxxx Fromuser=5804xxxx Secret=<we provided> Host=1511.zonetel.com Type=friend Insecure=very Qualify=no Nat=yes Canreinvite=no Disallow=all Allow=alaw Allow=ulaw Permit= / Sample elastix setup 1.Basic>Trunks>Add Trunk Outbound callerid = 5804xxxx Dial Rules =. Peer details= host=1511.zonetel.com type=peer permit= / Defaultexpiry=600 register string=5804xxxx:your sip pwd@1511.zonetel.com:5060/5804xxxx~600

4 2.Basic>Outbound Routes>Add Routes ;; if you dial 9 to call out via sip, etc. Dial Patterns= 9. Trunk Sequence=<the trunk created in 1> 3.Inbound Call Control>Inbound Routes>Add Incoming Route DID number=5804xxxx Extensions=<the extension to answer inbound call from 5804xxxx> Ports required Please check that your firewall allows below outgoing traffic to our network tcp&udp for sip signaling udp to / for calling HK All udp to / for calling IDD Caller ID Please ensure you are using our 5804xxxx as the callerid to our SIP trunk.

5 FAQ After i dial out, the destination phone rings, but we could not hear each other. Check that you don't have any firewall rules blocking traffic from/to our ip / Please check that your firewall allows udp traffic to your SIP server/endpoint. If you are using Asterisk, you can check its RTP port range in /etc/asterisk/rtp.conf. One way audio could be due to problems with NAT handling. Please check the nat and canreinvite parameters in your sip configuration. Please check that you are using g711 (alaw or ulaw). By default, we support g711. For g729 support, please make prior arrangement with our service team. How to dial to HK? Simply use 8-digits as the HK destination number (the '852' country prefix is not necessary). The callee will see your 5804xxxx as caller id. Outgoing calls to HK is free of charge. Please make sure your are using 5804xxxx as the callerid How to make international call? In our platform, we treat all non-hk calls as international call. Use dial string 1511+country+area+destination to make international calls. Call charges apply. Please contact us for rate plan. Please make sure you are using 5804xxxx as the callerid For security reason, we highly recommend IDD PIN. Please contact our service team for the procedures.

6 Sometimes, I could not receive incoming calls. It could be due to sip registration being expired too early. We recommend a sip registration TTL value of at least 600. Please check that 5804xxxx is being used by one sip end point only. (This applies when your sip end point directly registers to our server rather than your own PBX) I do not get a dial tone from zonetel Please check that your firewall permits traffic to ports of /27. I can dial to HK, but not other destinations Please allow all UDP ports from our subnet /27. The is for HK calling. We will use any other UDP ports for international calling. By default, 5804xxxx is IDD-disabled. Please contact our service team to open the IDD service and inquire the IDD rates. We highly recommend IDD PIN. Please contact our service team for the procedures. The voice quality is not good VOIP is always sensitive to bandwidth stability and network delay. A stable internet path from your site to our /27 is important for a good voice quality. You could also use G729 for voice call to reduce bandwidth demand. Note that G729 is not a free codec and you might need to purchase additional license in your end points or sip server in order to use G729. G729 is also a lossy codec (it is compressed and voice quality is not as good as G711) and is only suitable for voice. If you use 5804xxxx for fax, G711 must

7 be used I can dial out directly from phone, but call forward fails When your PBX forwards call via our SIP trunk, please ensure that it is sending 5804xxxx as the callerid. Outside caller resulted from a forwarded call will not be recognized by us. If you are using Elastix, please choose trunk CID options 'Block Foreign CID' or 'Force Trunk CID'. Still have problem? Please send us the following information for troubleshooting. Sip.conf Sip show peer <peer name> Session log after you turned on sip debug 'sip set debug on'