no voice when calling another phone

Get help with installing, upgrading and running Asterisk.

Moderators: muppetmaster, Moderator, Support

no voice when calling another phone

Postby chaoschef » Fri Aug 23, 2013 8:09 pm

Hi,

For some reason I can not hear anything when i call from my ip phone to my softphone.
Ip phone to asterisk answering machine works
soft phone to asterisk answering machine works
and i can hear my voice when i use echo.
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby ambiorixg12 » Fri Aug 23, 2013 9:14 pm

Post the cli ouput when you make a call . Check codecs too, and setting on both phones.
ambiorixg12
Astmaster
 
Posts: 967
Joined: Sun Mar 04, 2007 9:32 pm
Location: Dominican Republic

Re: no voice when calling another phone

Postby chaoschef » Fri Aug 23, 2013 9:45 pm

Code: Select all
  == Using SIP RTP CoS mark 5
    -- Executing [101@home:1] Dial("SIP/101-00000008", "SIP/ipphone") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ipphone
    -- SIP/ipphone-00000009 is ringing
    -- SIP/ipphone-00000009 answered SIP/101-00000008
    -- Locally bridging SIP/101-00000008 and SIP/ipphone-00000009
  == Spawn extension (home, 101, 1) exited non-zero on 'SIP/101-00000008'


im using in sip.conf

disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
allow=h261
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby ambiorixg12 » Fri Aug 23, 2013 10:01 pm

every thing look fine, have you try with different s softphones. ZOIPER, XLITE
ambiorixg12
Astmaster
 
Posts: 967
Joined: Sun Mar 04, 2007 9:32 pm
Location: Dominican Republic

Re: no voice when calling another phone

Postby chaoschef » Fri Aug 23, 2013 10:45 pm

yea just tried the other softphones, still same result.
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby navaismo » Fri Aug 23, 2013 10:58 pm

Show us the sip debug of a failed call
navaismo
Salt of the Asterisk
 
Posts: 1610
Joined: Mon Dec 07, 2009 1:30 pm
Location: Mexico City, Mexico

Re: no voice when calling another phone

Postby chaoschef » Fri Aug 23, 2013 11:16 pm

Code: Select all
<--- Reliably Transmitting (no NAT) to 192.168.0.126:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.126:5060;branch=z9hG4bK-d8754z-8cd78d898fc62bf5-1---d8754z-;received=192.168.0.126
From: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
To: <sip:101@192.168.0.150;transport=UDP>;tag=as60f65b48
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 1 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12a533a7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.126:5060 --->
ACK sip:101@192.168.0.150;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.126:5060;branch=z9hG4bK-d8754z-8cd78d898fc62bf5-1---d8754z-
Max-Forwards: 70
To: <sip:101@192.168.0.150;transport=UDP>;tag=as60f65b48
From: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.126:5060 --->
INVITE sip:101@192.168.0.150;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.126:5060;branch=z9hG4bK-d8754z-efed6d5e894b82a0-1---d8754z-
Max-Forwards: 70
Contact: <sip:101@192.168.0.126:5060;transport=UDP>
To: <sip:101@192.168.0.150;transport=UDP>
From: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.39 r16838
Authorization: Digest username="101",realm="asterisk",nonce="12a533a7",uri="sip:101@192.168.0.150;transport=UDP",response="49297339b0d75fb3b1a2dedba8036a91",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 329

v=0
o=Zoiper_user 0 0 IN IP4 192.168.0.126
s=Zoiper_session
c=IN IP4 192.168.0.126
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.0.126:5060 (no NAT)
Using INVITE request as basis request - NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
Found peer '101' for '101' from 192.168.0.126:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4040e (gsm|ulaw|alaw|ilbc|h261), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.126:8000
Looking for 101 in home (domain 192.168.0.150)
list_route: hop: <sip:101@192.168.0.126:5060;transport=UDP>

<--- Transmitting (no NAT) to 192.168.0.126:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.126:5060;branch=z9hG4bK-d8754z-efed6d5e894b82a0-1---d8754z-;received=192.168.0.126
From: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
To: <sip:101@192.168.0.150;transport=UDP>
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:101@192.168.0.150:5060>
Content-Length: 0


<------------>
    -- Executing [101@home:1] Dial("SIP/101-00000014", "SIP/ipphone") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 26760
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.157:5060:
INVITE sip:ipphone@192.168.0.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK0fa4c2be
Max-Forwards: 70
From: "101" <sip:101@192.168.0.150>;tag=as7d8e9534
To: <sip:ipphone@192.168.0.157>
Contact: <sip:101@192.168.0.150:5060>
Call-ID: 4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r396427
Date: Sat, 24 Aug 2013 05:15:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 372

v=0
o=root 1605865195 1605865195 IN IP4 192.168.0.150
s=Asterisk PBX SVN-branch-1.8-r396427
c=IN IP4 192.168.0.150
t=0 0
m=audio 26760 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/ipphone

<--- SIP read from UDP:192.168.0.157:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK0fa4c2be
From: "101" <sip:101@192.168.0.150>;tag=as7d8e9534
To: <sip:ipphone@192.168.0.157>
Call-ID: 4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060
CSeq: 102 INVITE
User-Agent: Grandstream BT110 1.0.8.33
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.157:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK0fa4c2be
From: "101" <sip:101@192.168.0.150>;tag=as7d8e9534
To: <sip:ipphone@192.168.0.157>;tag=204a248a5e99655d
Call-ID: 4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060
CSeq: 102 INVITE
User-Agent: Grandstream BT110 1.0.8.33
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
list_route: no route
    -- SIP/ipphone-00000015 is ringing

<--- Transmitting (no NAT) to 192.168.0.126:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.126:5060;branch=z9hG4bK-d8754z-efed6d5e894b82a0-1---d8754z-;received=192.168.0.126
From: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
To: <sip:101@192.168.0.150;transport=UDP>;tag=as42a41c35
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:101@192.168.0.150:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.0.157:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK0fa4c2be
From: "101" <sip:101@192.168.0.150>;tag=as7d8e9534
To: <sip:ipphone@192.168.0.157>;tag=204a248a5e99655d
Call-ID: 4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060
CSeq: 102 INVITE
User-Agent: Grandstream BT110 1.0.8.33
Contact: <sip:ipphone@192.168.0.157>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 160

v=0
o=ipphone 8000 8000 IN IP4 192.168.0.157
s=SIP Call
c=IN IP4 192.168.0.157
t=0 0
m=audio 5004 RTP/AVP 0
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4040e (gsm|ulaw|alaw|ilbc|h261), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.157:5004
list_route: hop: <sip:ipphone@192.168.0.157>
set_destination: Parsing <sip:ipphone@192.168.0.157> for address/port to send to
set_destination: set destination to 192.168.0.157:5060
Transmitting (no NAT) to 192.168.0.157:5060:
ACK sip:ipphone@192.168.0.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK0e04d10f
Max-Forwards: 70
From: "101" <sip:101@192.168.0.150>;tag=as7d8e9534
To: <sip:ipphone@192.168.0.157>;tag=204a248a5e99655d
Contact: <sip:101@192.168.0.150:5060>
Call-ID: 4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r396427
Content-Length: 0


---
    -- SIP/ipphone-00000015 answered SIP/101-00000014
Audio is at 8312
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.126:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.126:5060;branch=z9hG4bK-d8754z-efed6d5e894b82a0-1---d8754z-;received=192.168.0.126
From: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
To: <sip:101@192.168.0.150;transport=UDP>;tag=as42a41c35
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:101@192.168.0.150:5060>
Content-Type: application/sdp
Content-Length: 371

v=0
o=root 1372235183 1372235183 IN IP4 192.168.0.150
s=Asterisk PBX SVN-branch-1.8-r396427
c=IN IP4 192.168.0.150
t=0 0
m=audio 8312 RTP/AVP 0 8 98 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Locally bridging SIP/101-00000014 and SIP/ipphone-00000015

<--- SIP read from UDP:192.168.0.126:5060 --->
ACK sip:101@192.168.0.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.126:5060;branch=z9hG4bK-d8754z-d26a491e15b27589-1---d8754z-
Max-Forwards: 70
Contact: <sip:101@192.168.0.126:5060;transport=UDP>
To: <sip:101@192.168.0.150;transport=UDP>;tag=as42a41c35
From: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 2 ACK
User-Agent: Zoiper for Windows 2.39 r16838
Authorization: Digest username="101",realm="asterisk",nonce="12a533a7",uri="sip:101@192.168.0.150;transport=UDP",response="49297339b0d75fb3b1a2dedba8036a91",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.0.157:5060:
OPTIONS sip:ipphone@192.168.0.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK286a1f11
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.150>;tag=as75f5cbc9
To: <sip:ipphone@192.168.0.157>
Contact: <sip:asterisk@192.168.0.150:5060>
Call-ID: 7adc685a5bb12b3c3917d6e4562925cc@192.168.0.150:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-branch-1.8-r396427
Date: Sat, 24 Aug 2013 05:15:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.157:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK286a1f11
From: "asterisk" <sip:asterisk@192.168.0.150>;tag=as75f5cbc9
To: <sip:ipphone@192.168.0.157>;tag=204a248a5e99655d
Call-ID: 7adc685a5bb12b3c3917d6e4562925cc@192.168.0.150:5060
CSeq: 102 OPTIONS
User-Agent: Grandstream BT110 1.0.8.33
Contact: <sip:ipphone@192.168.0.157>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '7adc685a5bb12b3c3917d6e4562925cc@192.168.0.150:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.0.157:5060 --->
BYE sip:101@192.168.0.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157;branch=z9hG4bK1aca7a24a1c32077
From: <sip:ipphone@192.168.0.157>;tag=204a248a5e99655d
To: "101" <sip:101@192.168.0.150>;tag=as7d8e9534
Supported: replaces
Call-ID: 4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060
CSeq: 42909 BYE
User-Agent: Grandstream BT110 1.0.8.33
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.157:5060 (no NAT)
Scheduling destruction of SIP dialog '4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.157:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.157;branch=z9hG4bK1aca7a24a1c32077;received=192.168.0.157
From: <sip:ipphone@192.168.0.157>;tag=204a248a5e99655d
To: "101" <sip:101@192.168.0.150>;tag=as7d8e9534
Call-ID: 4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060
CSeq: 42909 BYE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (home, 101, 1) exited non-zero on 'SIP/101-00000014'
Scheduling destruction of SIP dialog 'NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:101@192.168.0.126:5060;transport=UDP> for address/port to send to
set_destination: set destination to 192.168.0.126:5060
Reliably Transmitting (no NAT) to 192.168.0.126:5060:
BYE sip:101@192.168.0.126:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK4e987512
Max-Forwards: 70
From: <sip:101@192.168.0.150;transport=UDP>;tag=as42a41c35
To: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 102 BYE
User-Agent: Asterisk PBX SVN-branch-1.8-r396427
Proxy-Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:192.168.0.150", nonce="", response="e512713261a212aa9dca2eee5676f0d8"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Retransmitting #1 (no NAT) to 192.168.0.126:5060:
BYE sip:101@192.168.0.126:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK4e987512
Max-Forwards: 70
From: <sip:101@192.168.0.150;transport=UDP>;tag=as42a41c35
To: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 102 BYE
User-Agent: Asterisk PBX SVN-branch-1.8-r396427
Proxy-Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:192.168.0.150", nonce="", response="e512713261a212aa9dca2eee5676f0d8"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.126:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK4e987512
Contact: <sip:101@192.168.0.126:5060;transport=UDP>
To: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
From: <sip:101@192.168.0.150;transport=UDP>;tag=as42a41c35
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 102 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.' Method: ACK

<--- SIP read from UDP:192.168.0.126:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK4e987512
Contact: <sip:101@192.168.0.126:5060;transport=UDP>
To: "101"<sip:101@192.168.0.150;transport=UDP>;tag=b343d144
From: <sip:101@192.168.0.150;transport=UDP>;tag=as42a41c35
Call-ID: NzQ0N2Y4NGYxNWMxMDRjMTVmZTFkZDFmZmJlZjMwZDE.
CSeq: 102 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.126:5060 --->


<------------->
Reliably Transmitting (no NAT) to 192.168.0.126:5060:
OPTIONS sip:101@192.168.0.126:5060;rinstance=0394aa0ba1cffc11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK0c5eca8e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.150>;tag=as656b16eb
To: <sip:101@192.168.0.126:5060;rinstance=0394aa0ba1cffc11;transport=UDP>
Contact: <sip:asterisk@192.168.0.150:5060>
Call-ID: 21cc11d66f4ba3de1fcdd6e6109f555c@192.168.0.150:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-branch-1.8-r396427
Date: Sat, 24 Aug 2013 05:15:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.126:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK0c5eca8e
Contact: <sip:192.168.0.126:5060>
To: <sip:101@192.168.0.126:5060;rinstance=0394aa0ba1cffc11;transport=UDP>;tag=83473000
From: "asterisk"<sip:asterisk@192.168.0.150>;tag=as656b16eb
Call-ID: 21cc11d66f4ba3de1fcdd6e6109f555c@192.168.0.150:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.39 r16838
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '21cc11d66f4ba3de1fcdd6e6109f555c@192.168.0.150:5060' Method: OPTIONS
Really destroying SIP dialog '4bca9abf6b4a21a473ca270e0fe4ddbb@192.168.0.150:5060' Method: BYE
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby navaismo » Sat Aug 24, 2013 12:19 am

Looks normal and no errors. Just noticed that you are using a branch version of 1.8 is any reason to not use the latest release of 1.8 or 11? Also check in the rtp debug that you see packets from both phone's IP.
navaismo
Salt of the Asterisk
 
Posts: 1610
Joined: Mon Dec 07, 2009 1:30 pm
Location: Mexico City, Mexico

Re: no voice when calling another phone

Postby ambiorixg12 » Sat Aug 24, 2013 3:47 pm

post your sip.conf setting for the 2 phones
ambiorixg12
Astmaster
 
Posts: 967
Joined: Sun Mar 04, 2007 9:32 pm
Location: Dominican Republic

Re: no voice when calling another phone

Postby chaoschef » Sat Aug 24, 2013 11:08 pm

sip.conf

Code: Select all
[general]
context=default         
allowguest=no                   
srvlookup=yes                   
udpbindaddr=0.0.0.0             
tcpenable=no                   
;videosupport=yes


disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
allow=h261




[ipphone]
type=friend
username=ipphone
secret=xxxxx
host=dynamic
context=home
canreinivte=no
nat=no
qualify=yes

[101]
type=friend
secret=xxxxxx
qualify=yes   
nat=no       
host=dynamic   
canreinvite=no
context=home
;port=5061     
user=101
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby chaoschef » Sun Aug 25, 2013 12:36 am

i installed asterisk 1.8.23.0 and still have the same problem with no voice between the phones, but i can hear my voice when i do an echo to asterisk.
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby chaoschef » Sun Aug 25, 2013 12:43 am

i used: rtp set debug on, and i didnt see any packets, but when i called asterisk from one of the phones there was a lot of packets being sent, have i missed something?
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby ambiorixg12 » Sun Aug 25, 2013 2:38 pm

Try with the directmedia option in your sip.conf file

By default, Asterisk tries to re-invite media streams to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
;
;directmedia=yes ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
ambiorixg12
Astmaster
 
Posts: 967
Joined: Sun Mar 04, 2007 9:32 pm
Location: Dominican Republic

Re: no voice when calling another phone

Postby chaoschef » Sun Aug 25, 2013 5:01 pm

ok i tried that, we must be getting close now.

I am getting voice going one way now.

The receiver of the call can hear voice and not send voice, and the sender of a call can send voice but not receive voice.
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby ambiorixg12 » Sun Aug 25, 2013 5:14 pm

You have some issue with the RTP traffic read on this document the part of MEDIA HANDLING --
ambiorixg12
Astmaster
 
Posts: 967
Joined: Sun Mar 04, 2007 9:32 pm
Location: Dominican Republic

Re: no voice when calling another phone

Postby ambiorixg12 » Sun Aug 25, 2013 5:24 pm

Last edited by ambiorixg12 on Sun Aug 25, 2013 6:14 pm, edited 2 times in total.
ambiorixg12
Astmaster
 
Posts: 967
Joined: Sun Mar 04, 2007 9:32 pm
Location: Dominican Republic

Re: no voice when calling another phone

Postby chaoschef » Sun Aug 25, 2013 5:52 pm

thank you for your help, i got it working now,

I used directmedia=outgoing for the ip phone
and canreinvite=no for both softphones.
chaoschef
Newsterisk
 
Posts: 12
Joined: Wed Aug 21, 2013 1:59 am

Re: no voice when calling another phone

Postby ambiorixg12 » Sun Aug 25, 2013 5:57 pm

canreinvite= was renamed to directmedia= in Asterisk 1.6.2 to more accurately describe what this setting does. See also the closely related setting directrtpsetup.

Asterisk 1.8 added directmediapermit and directmediadeny to limit which peers can send direct media to each other.

http://www.voip-info.org/wiki/view/Aste ... anreinvite
ambiorixg12
Astmaster
 
Posts: 967
Joined: Sun Mar 04, 2007 9:32 pm
Location: Dominican Republic


Return to Asterisk Support

Who is online

Users browsing this forum: No registered users and 1 guest