No Audio & disconnect on SOME outgoing calls

Asterisk Asterisk Support
1

Good afternoon everyone,

I would very much appreciate any suggestions or points anybody could give me with a very frustrating issue I’m experiencing with outgoing calls.

From what I can tell this issue suddenly began and no changes or updates had been made to our Asterisk server which lead me to thing it was the SIP trunk provider but they claim to not have made any changes and haven’t been able to resolve the issue for over a week now.

The Asterisk server in question is an AsteriskNow x64 install and is updated to Asterisk 11.14.0

The issue we are having is that SOME outgoing calls have no outgoing audio (we can hear them but they can’t hear us) and then the call is terminated after 15 seconds.

Everything I’ve read and googled points towards a NAT issue but the current setup is working and has been working for a while, we can receive calls without any issues and the majority of outgoing calls work.

Calls to 0860 numbers (who usually belong to medical aid, city call centres and bank call centres) and some 083 numbers (083 belongs to one of the 4 mobile service providers in South Africa) fail 95% of the time and every so often go through without an issue.
In the case of the 0860 numbers, they are usually answered by an IVR and pressing any keys as directed by the voice prompts are not transmitted (i.e. press 1 for sales, pressing 1 does nothing and the IVR continues) until 15 seconds later when the call is cut.

I’ve been watching the CLI and have noticed an interesting pattern:

[color=#00BF00]Successful call:[/color]

-- Called SIP/MTN/0723756899
    -- SIP/MTN-0000004b is making progress passing it to SIP/125-0000004a
       > 0x7f339c22f4d0 -- Probation passed - setting RTP source address to 10.10.2.204:16838
       > 0x7f33a0924510 -- Probation passed - setting RTP source address to 192.168.3.125:5004
    -- SIP/MTN-0000004b is ringing
    -- SIP/MTN-0000004b is ringing
    -- SIP/MTN-0000004b answered SIP/125-0000004a

[color=#00BF00]Successful call:[/color]

-- Called SIP/MTN/1026
    -- SIP/MTN-00000051 is making progress passing it to SIP/125-00000050
       > 0x7f339c22f4d0 -- Probation passed - setting RTP source address to 10.10.2.204:16852
       > 0x7f33a0ca7670 -- Probation passed - setting RTP source address to 192.168.3.125:5004
    -- SIP/MTN-00000051 is ringing
    -- SIP/MTN-00000051 answered SIP/125-00000050

[color=#FF0000]Failed Call:[/color]

 -- Called SIP/MTN/0860123000
    -- SIP/MTN-0000004d is making progress passing it to SIP/125-0000004c
       > 0x7f33a0ca7670 -- Probation passed - setting RTP source address to 192.168.3.125:5004
       > 0x7f339c22f4d0 -- Probation passed - setting RTP source address to 10.10.2.204:1999
       > 0x7f339c22f4d0 -- Probation passed - setting RTP source address to 10.10.2.204:1999
    -- SIP/MTN-0000004d is ringing
    -- SIP/MTN-0000004d is ringing
    -- SIP/MTN-0000004d answered SIP/125-0000004c

[color=#FF0000]Failed Call:[/color]

-- Called SIP/MTN/0860562874
    -- SIP/MTN-0000003c is making progress passing it to SIP/125-0000003b
       > 0x7f33a0d8ee90 -- Probation passed - setting RTP source address to 192.168.3.125:5004
       > 0x7f339c22f4d0 -- Probation passed - setting RTP source address to 10.10.2.204:1996
       > 0x7f339c22f4d0 -- Probation passed - setting RTP source address to 10.10.2.204:1996
    -- SIP/MTN-0000003c is ringing
    -- SIP/MTN-0000003c answered SIP/125-0000003b

Every failed call always has 2 [color=#BF0000]Probation passed - setting RTP source address to 10.10.2.204[/color] lines whereas all successful calls only have 1.
Secondly, all failed calls are always “setting the RTP source address to” 10.10.2.204 [SOME PORT LOWER THAN 10000] whereas all successful calls always set the port to a port between 10000 and 20000.

I have also generated masses of calls logs which I am more than happy to supply if required but I didn’t want to make this post miles long.

Am I doing something wrong? Or is this the fault of the SIP provider?

Thanks!

2

Hey all,

I just wanted to check if anybody has any ideas or comments?
I can post more information or logs/call traces if it would help?

3

I am having the same problem. Hope some one can help .

4

Hi,

I encountered a similar issue in regards to the RTP ports for handsets being out of the default Asterisk range.

The brand of handsets were Polycom SP IP 331. After reading the manual, I discovered the RTP port range started at 2222. Per the manual, I modified the sip.config to include the following:

Example:

<port> <RTP tcpIpApp.port.rtp.filterByIp="{$filterByIp|1}" tcpIpApp.port.rtp.filterByPort="{$filterByPort|0}" tcpIpApp.port.rtp.mediaPortRangeStart="10024"/> </port>

I hope this helps.

_Zach T (Netflash)

5

That’s irrelevant. It is perfectly valid to have non-overlapping port ranges for the two sides. Each side chooses which port to use at its end, and communicates its choice to the other end.