[solved] IAX2--Asterisk--SIP trunk CCM--SIP = some problems

Dear all.

I am new to voip and need some advice/explanation.

One of my soft phone Yate connects through VPN to asterisk, it uses IAX2.
Asterisk has SIP trunk to CMM.
Another end point is Polycom SIP phone, and it uses SIP to register in CCM.

I am experiencing some problems, but want to solve them in sequence.

First:
When IAX2 yate client calls to Polycom SIP, no one hear the ring. There is no ring sound on Polycom (but there is indication on phone screen that line 1 is in use) and yate soft phone doesn’t ring.
If the Polycom user answers the call, normal conversation will start.

If yate uses SIP accont with asterist - everything is OK.

Thanks for answers.

Problem was with Yate and Polycom phones.

Cisco phones and sflphone works.

Thanks.

OK, I need advice, really.

What I have: polycom SP320 registered on CMM. I have software IAX2 client registered on Asterisk.
I call from softphone to polycom and Asterisk dial it by sip name without IP.
Problem - polycom doesn’t ring, although it indicates that line is using. My soft client also doesn’t make ring sound. If polycom answers, we can talk. Main problem here is plyconm doesn’t ring.
Messages from asterisk:

Reliably Transmitting (no NAT) to 172.16.248.161:5060: -----> to the 1 CCM
INVITE sip:01119@172.16.248.161 SIP/2.0
Via: SIP/2.0/UDP 172.16.248.139:5060;branch=z9hG4bK7eff77f5
Max-Forwards: 70
From: “530010” sip:530010@172.16.248.139;tag=as4009ba52
To: sip:01119@172.16.248.161
Contact: sip:530010@172.16.248.139:5060

Reliably Transmitting (no NAT) to 172.16.248.162:5060: -----> to the 2 CCM
INVITE sip:01119@172.16.248.162 SIP/2.0
Via: SIP/2.0/UDP 172.16.248.139:5060;branch=z9hG4bK1c81b2de
Max-Forwards: 70
From: “530010” sip:530010@172.16.248.139;tag=as7e1a7fb8
To: sip:01119@172.16.248.162
Contact: sip:530010@172.16.248.139:5060

<— SIP read from UDP:172.16.248.161:5060 —>
SIP/2.0 100 Trying
Date: Wed, 01 May 2013 19:25:14 GMT
From: “530010” sip:530010@172.16.248.139;tag=as4009ba52
Allow-Events: presence
Content-Length: 0
To: sip:01119@172.16.248.161

<— SIP read from UDP:172.16.248.162:5060 —>
SIP/2.0 100 Trying
Date: Wed, 01 May 2013 19:25:14 GMT
From: “530010” sip:530010@172.16.248.139;tag=as7e1a7fb8
Allow-Events: presence
Content-Length: 0
To: sip:01119@172.16.248.162

<— SIP read from UDP:172.16.248.161:5060 —>
SIP/2.0 180 Ringing
Date: Wed, 01 May 2013 19:25:14 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: “530010” sip:530010@172.16.248.139;tag=as4009ba52

<— SIP read from UDP:172.16.248.162:5060 —>
SIP/2.0 180 Ringing
Date: Wed, 01 May 2013 19:25:14 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: “530010” sip:530010@172.16.248.139;tag=as7e1a7fb8

Second way: Asterisk dial to polycom by SIP/510616@10.1.3.xxx.
In this case polycom is ringing, my iax2 client isn’t.

I don’t care about my client. I need to know why polycoms aren’t ringing, because I don’t know their IPs.

Thanks for any suggestions.

Thank you for the silence.

Now is 2013 and linux doesn’t have any normal IAX client.

I found Zoiper and it works properly. Phones are ringing.

Thanks again.