# Problem with sound in external sip calls

**URL:** <https://community.asterisk.org/t/problem-with-sound-in-external-sip-calls/28438>\
**Category:** Asterisk Support\
**Created:** [August 13, 2009, 10:01am UTC](https://community.asterisk.org/t/problem-with-sound-in-external-sip-calls/28438 "2009-08-13T10:01:36Z")\
**Posts on this page:** 5\
**Page:** 1

<div class="post-metadata">

**Author:** ![iakovosmavro](https://avatars.discourse-cdn.com/v4/letter/i/f1d935/32.png) [@iakovosmavro](https://community.asterisk.org/u/iakovosmavro)\
**Post date:** [August 13, 2009, 10:01am UTC](https://community.asterisk.org/t/problem-with-sound-in-external-sip-calls/28438/1 "2009-08-13T10:01:36Z")

</div>

Hi,

I am having a problem with sound. I have tried almost everything but it looks like the problem is in asterisk.

Configuration:  
First off, I use  
asterisk-1.4.20  
freepbx-2.3.1  
CentOS 5.1  
I have connected asterisk with an external voip provider, so I can receive and place calls through the voip server. When I call from an extension to my cell phone or when I call from my cell phone to an extension I don’t have any sound problem.

Description of the problem:  
I have configured freepbx to forward all incoming calls to a ring group that calls my cell phone. In this way, when somebody calls my voip number (let’s say A) my cell phone rings (let’s say cell phone number B). When I pick up the cell phone there is no sound.

I have used wireshark and I can see that there are no RTP packets transfered between the voip provider and the linux machine. I have made several tests and the problem happens only when number A and number B are going through the same voip provider. For example, in my case my voip provider connects to my linux machine when somebody calls me (call A) and the linux machine connects back to my voip provider to place call B.

I have attached the log from asterisk in the end. I think asterisk is getting confused when a sip call is initiated from and forwarded to the same sip provider. In this log I think the problem is close to the end in line:  
"To: sip:0858780396@127.0.0.1;user=phone;tag=as04e1b9f4"  
The IP=127.0.0.1 doesn’t look ok to me but I am not sure if this is the problem.

Please let me know how I can solve this problem or if you need more info.

Thanks in advance,  
Iakovos

Asterisk Log:  
GotoIf(“Local/900302810226134@from-internal-917e,2”, “0?bypass|1”) in new stack  
– Executing [s@macro-dialout-trunk:24] GotoIf(“Local/900302810226134@from-internal-917e,2”, “0?customtrunk”) in new stack  
– Executing [s@macro-dialout-trunk:25] Dial(“Local/900302810226134@from-internal-917e,2”, “SIP/xs4all/00302810226134|300|TW”) in new stackAudio is at 192.168.2.5 port 10560  
Adding codec 0x8 (alaw) to SDP  
Adding codec 0x2 (gsm) to SDP  
Adding codec 0x4 (ulaw) to SDP  
Reliably Transmitting (NAT) to 82.101.62.99:5060:  
INVITE [sip:00302810226134@sip.xs4all.nl](mailto:sip:00302810226134@sip.xs4all.nl) SIP/2.0  
Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bK125389e4;rport  
From: “00302810241648” sip:0858780396@sip.xs4all.nl;tag=as0a6974dd  
To: sip:00302810226134@sip.xs4all.nl  
Contact: sip:0858780396@192.168.2.5  
Call-ID: [1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl](mailto:1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl)  
CSeq: 102 INVITE  
User-Agent: Asterisk PBX  
Max-Forwards: 70  
Date: Thu, 13 Aug 2009 09:37:35 GMT  
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY  
Supported: replaces  
Content-Type: application/sdp  
Content-Length: 229

v=0  
o=root 10876 10876 IN IP4 192.168.2.5  
s=session  
c=IN IP4 192.168.2.5  
t=0 0  
m=audio 10560 RTP/AVP 8 3 0  
a=rtpmap:8 PCMA/8000  
a=rtpmap:3 GSM/8000  
a=rtpmap:0 PCMU/8000  
a=silenceSupp:off - - - -  
a=ptime:20  
a=sendrecv

— – Called xs4all/00302810226134olo\*CLI\>  
\<— SIP read from 82.101.62.99:5060 —\>  
SIP/2.0 100 Giving a try  
Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bK125389e4;rport=5061;received=94.71.126.12  
From: “00302810241648” sip:0858780396@sip.xs4all.nl;tag=as0a6974dd  
To: sip:00302810226134@sip.xs4all.nl  
Call-ID: [1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl](mailto:1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl)  
CSeq: 102 INVITE  
Server: OpenSIPS (1.4.4-notls (i386/linux))  
Content-Length: 0

\<-------------\>— (8 headers 0 lines) —olo\*CLI\>  
\<— SIP read from 82.101.62.99:5060 —\>  
SIP/2.0 407 authentication required  
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE  
Call-ID: [1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl](mailto:1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl)  
Contact: sip:00302810226134@82.101.62.99:5060;user=phone  
CSeq: 102 INVITE  
From: “00302810241648” sip:0858780396@sip.xs4all.nl;tag=as0a6974dd  
Proxy-Authenticate: Digest realm=“[sip.xs4all.nl](http://sip.xs4all.nl)”,nonce=“12fe379265da7acd1f50de12768a0bac”,opaque=“12fc27bb1525c74”,stale=false,algorithm=MD5  
Record-Route: sip:82.101.62.115;lr;r2=on;ftag=as0a6974dd,sip:82.101.63.5;lr;r2=on;ftag=as0a6974dd  
Server: Cirpack/v4.41b (gw\_sip)  
To: sip:00302810226134@sip.xs4all.nl;tag=00-07714-12fe3878-6b4127447  
Via: SIP/2.0/UDP 192.168.2.5:5060;received=94.71.126.12;rport=5061;branch=z9hG4bK125389e4  
Content-Length: 0

\<-------------\>— (12 headers 0 lines) —Transmitting (NAT) to 82.101.62.99:5060:  
ACK [sip:00302810226134@sip.xs4all.nl](mailto:sip:00302810226134@sip.xs4all.nl) SIP/2.0  
Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bK125389e4;rport  
From: “00302810241648” sip:0858780396@sip.xs4all.nl;tag=as0a6974dd  
To: sip:00302810226134@sip.xs4all.nl;tag=00-07714-12fe3878-6b4127447  
Contact: sip:0858780396@192.168.2.5  
Call-ID: [1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl](mailto:1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl)  
CSeq: 102 ACK  
User-Agent: Asterisk PBX  
Max-Forwards: 70  
Content-Length: 0

—Audio is at 192.168.2.5 port 10560Adding codec 0x8 (alaw) to SDPAdding codec 0x2 (gsm) to SDPAdding codec 0x4 (ulaw) to SDPReliably Transmitting (NAT) to 82.101.62.99:5060:  
INVITE [sip:00302810226134@sip.xs4all.nl](mailto:sip:00302810226134@sip.xs4all.nl) SIP/2.0  
Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bK1ef22546;rport  
From: “00302810241648” sip:0858780396@sip.xs4all.nl;tag=as0a6974dd  
To: sip:00302810226134@sip.xs4all.nl  
Contact: sip:0858780396@192.168.2.5  
Call-ID: [1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl](mailto:1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl)  
CSeq: 103 INVITE  
User-Agent: Asterisk PBX  
Max-Forwards: 70  
Proxy-Authorization: Digest username=“0858780396”, realm=“[sip.xs4all.nl](http://sip.xs4all.nl)”, algorithm=MD5, [uri="sip:00302810226134@sip.xs4all.nl](mailto:uri=%22sip:00302810226134@sip.xs4all.nl)", nonce=“12fe379265da7acd1f50de12768a0bac”, response=“b6bfac1571a9cc788791515931f46303”, opaque="12fc27bb1525c74"  
Date: Thu, 13 Aug 2009 09:37:35 GMT  
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY  
Supported: replaces  
Content-Type: application/sdp  
Content-Length: 229

v=0  
o=root 10876 10877 IN IP4 192.168.2.5  
s=session  
c=IN IP4 192.168.2.5  
t=0 0  
m=audio 10560 RTP/AVP 8 3 0  
a=rtpmap:8 PCMA/8000  
a=rtpmap:3 GSM/8000  
a=rtpmap:0 PCMU/8000  
a=silenceSupp:off - - - -  
a=ptime:20  
a=sendrecv

—olo\*CLI\>  
\<— SIP read from 82.101.62.99:5060 —\>  
SIP/2.0 100 Giving a try  
Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bK1ef22546;rport=5061;received=94.71.126.12  
From: “00302810241648” sip:0858780396@sip.xs4all.nl;tag=as0a6974dd  
To: sip:00302810226134@sip.xs4all.nl  
Call-ID: [1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl](mailto:1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl)  
CSeq: 103 INVITE  
Server: OpenSIPS (1.4.4-notls (i386/linux))  
Content-Length: 0

\<-------------\>— (8 headers 0 lines) —Really destroying SIP dialog ‘28af5fd304d2a31a4acaccdf3e091758@194.30.193.119’ Method: OPTIONSReally destroying SIP dialog '054140295bf7120278c7b8bf408665cb@vtrip … ateway.com’ Method: REGISTERolo\*CLI\>  
\<— SIP read from 82.101.62.99:5060 —\>  
SIP/2.0 200 OK  
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE  
Call-ID: [1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl](mailto:1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl)  
Contact: sip:82.101.62.99:5060  
Content-Type: application/sdp  
CSeq: 103 INVITE  
From: “00302810241648” sip:0858780396@sip.xs4all.nl;tag=as0a6974dd  
Record-Route: sip:82.101.62.115;lr;r2=on;ftag=as0a6974dd,sip:82.101.63.5;lr;r2=on;ftag=as0a6974dd  
Server: Cirpack/v4.41b (gw\_sip)  
To: sip:00302810226134@sip.xs4all.nl;tag=00-07696-12fe3887-05349abb7  
Via: SIP/2.0/UDP 192.168.2.5:5060;received=94.71.126.12;rport=5061;branch=z9hG4bK1ef22546  
Content-Length: 209

v=0  
o=cp10 125015626065 125015626065 IN IP4 82.101.42.134  
s=SIP Call  
c=IN IP4 194.109.8.3  
t=0 0  
m=audio 30426 RTP/AVP 8 0  
b=AS:64  
a=rtpmap:8 PCMA/8000/1  
a=rtpmap:0 PCMU/8000/1  
a=ptime:20  
a=sendrecv

\<-------------\>— (12 headers 11 lines)  
—Found RTP audio format 8Found RTP audio format 0  
Peer audio RTP is at port 194.109.8.3:30426  
Found audio description format PCMA for ID 8  
Found audio description format PCMU for ID 0  
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)  
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)  
Peer audio RTP is at port 194.109.8.3:30426  
list\_route: hop: sip:82.101.63.5;lr;r2=on;ftag=as0a6974ddlist\_route: hop: sip:82.101.62.115;lr;r2=on;ftag=as0a6974ddset\_destination: Parsing sip:82.101.63.5;lr;r2=on;ftag=as0a6974dd for address/port to send toset\_destination: set destination to 82.101.63.5, port 5060Transmitting (NAT) to 82.101.62.99:5060:  
ACK sip:82.101.62.99:5060 SIP/2.0  
Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bK7a1618fc;rport  
Route: sip:82.101.63.5;lr;r2=on;ftag=as0a6974dd,sip:82.101.62.115;lr;r2=on;ftag=as0a6974dd  
From: “00302810241648” sip:0858780396@sip.xs4all.nl;tag=as0a6974dd  
To: sip:00302810226134@sip.xs4all.nl;tag=00-07696-12fe3887-05349abb7  
Contact: sip:0858780396@192.168.2.5  
Call-ID: [1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl](mailto:1c023b89513fbdd56b45a8cb7ececa1c@sip.xs4all.nl)  
CSeq: 103 ACK  
User-Agent: Asterisk PBX  
Max-Forwards: 70  
Content-Length: 0

* * *

```
-- SIP/xs4all-086b9e88 answered Local/900302810226134@from-internal-917e,2
-- Local/900302810226134@from-internal-917e,1 answered SIP/0858780396-08757af8
-- Executing [s@macro-auto-blkvm:1] Set("Local/900302810226134@from-internal-917e,1", "__MACRO_RESULT=") in new stack
-- Executing [s@macro-auto-blkvm:2] DBdel("Local/900302810226134@from-internal-917e,1", "BLKVM/77100/SIP/0858780396-08757af8") in new stack
-- DBdel: family=BLKVM, key=77100/SIP/0858780396-08757af8

```

Audio is at 192.168.2.5 port 15246  
Adding codec 0x8 (alaw) to SDP  
Adding codec 0x4 (ulaw) to SDP  
\<— Reliably Transmitting (NAT) to 82.101.62.99:5060 —\>  
SIP/2.0 200 OK  
Via: SIP/2.0/UDP 82.101.63.5;branch=z9hG4bKf10e.bb07e2b6.0;received=82.101.62.99  
Via: SIP/2.0/UDP 82.101.62.99:5060;rport=5060;received=82.101.62.99;branch=z9hG4bK-1D86-2BF240E  
Record-Route: sip:82.101.63.5;r2=on;lr=on;ftag=28106-LT-12fe3852-388dc4173  
Record-Route: sip:82.101.62.115;r2=on;lr=on;ftag=28106-LT-12fe3852-388dc4173  
From: “00302810241648” sip:00302810241648@sip.xs4all.nl;user=phone;tag=28106-LT-12fe3852-388dc4173  
To: sip:0858780396@127.0.0.1;user=phone;tag=as04e1b9f4  
Call-ID: [28106-UX-12fe3851-13606d4b4@sip.xs4all.nl](mailto:28106-UX-12fe3851-13606d4b4@sip.xs4all.nl)  
CSeq: 316099941 INVITE  
User-Agent: Asterisk PBX  
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY  
Supported: replaces  
Contact: sip:s@192.168.2.5  
Content-Type: application/sdp  
Content-Length: 206

v=0  
o=root 10876 10876 IN IP4 192.168.2.5  
s=session  
c=IN IP4 192.168.2.5  
t=0 0  
m=audio 15246 RTP/AVP 8 0  
a=rtpmap:8 PCMA/8000  
a=rtpmap:0 PCMU/8000  
a=silenceSupp:off - - - -  
a=ptime:20  
a=sendrecv

\<------------\>olo\*CLI\>  
\<— SIP read from 82.101.62.99:5060 —\>  
ACK sip:s@94.71.126.12:5061 SIP/2.0  
Record-Route: sip:82.101.63.5;r2=on;lr=on;ftag=28106-LT-12fe3852-388dc4173  
Record-Route: sip:82.101.62.115;r2=on;lr=on;ftag=28106-LT-12fe3852-388dc4173  
Call-ID: [28106-UX-12fe3851-13606d4b4@sip.xs4all.nl](mailto:28106-UX-12fe3851-13606d4b4@sip.xs4all.nl)  
Contact: sip:82.101.62.99:5060  
CSeq: 316099941 ACK  
From: “00302810241648” sip:00302810241648@sip.xs4all.nl;user=phone;tag=28106-LT-12fe3852-388dc4173  
Max-Forwards: 30  
To: sip:0858780396@127.0.0.1;user=phone;tag=as04e1b9f4  
User-Agent: Cirpack/v4.41b (gw\_sip)  
Via: SIP/2.0/UDP 82.101.63.5;branch=z9hG4bKf10e.bb07e2b6.2  
Via: SIP/2.0/UDP 82.101.62.99:5060;rport=5060;received=82.101.62.99;branch=z9hG4bK-621B-2BF247A  
Content-Length: 0  
P-hint: rr-enforced

\<-------------\>— (14 headers 0 lines) —

---

<div class="post-metadata">

**Author:** ![ianplain](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ianplain/32/1776_2.png) [@ianplain](https://community.asterisk.org/u/ianplain)\
**Post date:** [August 13, 2009, 10:31am UTC](https://community.asterisk.org/t/problem-with-sound-in-external-sip-calls/28438/2 "2009-08-13T10:31:27Z")

</div>

Hi

Set canreinvite=no in the sip.conf

looks like the rtp may be reinviting

Ian

---

<div class="post-metadata">

**Author:** ![iakovosmavro](https://avatars.discourse-cdn.com/v4/letter/i/f1d935/32.png) [@iakovosmavro](https://community.asterisk.org/u/iakovosmavro)\
**Post date:** [August 13, 2009, 10:44am UTC](https://community.asterisk.org/t/problem-with-sound-in-external-sip-calls/28438/3 "2009-08-13T10:44:19Z")

</div>

Hi,

Thanks for the reply. It is already set. This is the configuration in sip.conf:

[xs4all]  
username=XXXX  
type=peer  
secret=XXXX  
qualify=yes  
nat=yes  
insecure=very  
[host=sip.xs4all.nl](http://host=sip.xs4all.nl)  
fromuser=XXXX  
[fromdomain=sip.xs4all.nl](http://fromdomain=sip.xs4all.nl)  
dtmfmode=inband  
disallow=all  
canreinvite=no  
allow=gsm  
allow=alaw  
allow=ulaw  
call-limit=50

thanks,  
Iakovos

---

<div class="post-metadata">

**Author:** ![iakovosmavro](https://avatars.discourse-cdn.com/v4/letter/i/f1d935/32.png) [@iakovosmavro](https://community.asterisk.org/u/iakovosmavro)\
**Post date:** [August 19, 2009, 1:56pm UTC](https://community.asterisk.org/t/problem-with-sound-in-external-sip-calls/28438/4 "2009-08-19T13:56:08Z")

</div>

I installed asterisk 1.4.26.1 but the problem remains the same. I don’t know what to try next.  
Please advise!

thanks,  
Iakovos

---

<div class="post-metadata">

**Author:** ![cluster37](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/cluster37/32/550_2.png) [@cluster37](https://community.asterisk.org/u/cluster37)\
**Post date:** [September 4, 2009, 5:00pm UTC](https://community.asterisk.org/t/problem-with-sound-in-external-sip-calls/28438/5 "2009-09-04T17:00:44Z")

</div>

Hello,

I am having the same problem after moving from asterisk 1.4 to asterisk 1.6.0. So I was wondering if you finally solved your problem, yes if yes how did you managed that?  
Thanks