Re: [AFMUG] OT SIP issue

[Lewis Bergman]

When all else fails, try tcp instead of udp.

On Sun, Apr 8, 2018, 5:39 PM Faisal Imtiaz  wrote:

> Sorry.. saw that was already asked.
>
> Another item that will break audio is if there is a codec mis-match or
> codec is not enabled ...
>
> Regards.
>
> Faisal Imtiaz
> Snappy Internet & Telecom
> http://www.snappytelecom.net
>
> Tel: 305 663 5518 x 232
>
> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net
>
> --
>
> *From: *"Faisal Imtiaz" 
> *To: *af@afmug.com
> *Sent: *Sunday, April 8, 2018 6:35:46 PM
>
>
> *Subject: *Re: [AFMUG] OT SIP issue
>
> not sure what you are doing.. but do you have can re-invite on ?
>
>
>
> Faisal Imtiaz
> Snappy Internet & Telecom
> http://www.snappytelecom.net
>
> Tel: 305 663 5518 x 232
>
> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net
>
> --------------
>
> *From: *"Chuck McCown" 
> *To: *af@afmug.com
> *Sent: *Sunday, April 8, 2018 2:27:28 PM
> *Subject: *[AFMUG] OT SIP issue
>
> Pulling our hair out.  The Aastra phones will call each other, we can send
> dtmf but no audio.  Linksys sip ata does the same thing.  These were
> working, very frustrating.
>
>

Re: [AFMUG] OT SIP issue

[Faisal Imtiaz]

Sorry.. saw that was already asked. 

Another item that will break audio is if there is a codec mis-match or codec is 
not enabled ... 

Regards. 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: "Faisal Imtiaz" 
> To: af@afmug.com
> Sent: Sunday, April 8, 2018 6:35:46 PM
> Subject: Re: [AFMUG] OT SIP issue

> not sure what you are doing.. but do you have can re-invite on ?

> Faisal Imtiaz
> Snappy Internet & Telecom
> http://www.snappytelecom.net

> Tel: 305 663 5518 x 232

> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>> From: "Chuck McCown" 
>> To: af@afmug.com
>> Sent: Sunday, April 8, 2018 2:27:28 PM
>> Subject: [AFMUG] OT SIP issue

>> Pulling our hair out. The Aastra phones will call each other, we can send 
>> dtmf
>> but no audio. Linksys sip ata does the same thing. These were working, very
>> frustrating.

Re: [AFMUG] OT SIP issue

[Faisal Imtiaz]

not sure what you are doing.. but do you have can re-invite on ? 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: "Chuck McCown" 
> To: af@afmug.com
> Sent: Sunday, April 8, 2018 2:27:28 PM
> Subject: [AFMUG] OT SIP issue

> Pulling our hair out. The Aastra phones will call each other, we can send dtmf
> but no audio. Linksys sip ata does the same thing. These were working, very
> frustrating.

Re: [AFMUG] OT SIP issue

[David Thake]

Have you run a pcap at both ends?

Sent from my iPad

> On Apr 9, 2018, at 12:24 AM, Paul Stewart  wrote:
> 
> I would add that when SIP ALG is on with some routers it breaks things too … 
> point being to try it both ways ☺
>  
> From: Af  on behalf of George Skorup 
> 
> Reply-To: 
> Date: Sunday, April 8, 2018 at 5:29 PM
> To: 
> Subject: Re: [AFMUG] OT SIP issue
>  
> What kind of router/firewall are you working with? No audio is usually a SIP 
> ALG thing. You need the ALG on to rewrite the SIP headers when behind NAT.
> 
> On 4/8/2018 2:30 PM, Chuck McCown wrote:
> Tried both ways, no joy.
> 
> Sent from my iPhone
> 
> On Apr 8, 2018, at 1:23 PM, Forrest Christian (List Account) 
>  wrote:
> 
> Are they behind nat?
>  
> Sounds like it might be a reinvite issue, asterisk will try to get out of the 
> audio path by telling the endpoints to talk directly to each other.   If nat 
> is involved asterisk will often tell the endpoints to talk directly even if 
> they have no direct connection between them.
>  
> Disabling reinvite may help if this is the case.
>  
> On Sun, Apr 8, 2018, 11:27 AM Chuck McCown  wrote:
> Pulling our hair out.  The Aastra phones will call each other, we can send 
> dtmf but no audio.  Linksys sip ata does the same thing.  These were working, 
> very frustrating. 
> 
> 

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Re: [AFMUG] OT SIP issue

[Paul Stewart]

I would add that when SIP ALG is on with some routers it breaks things too … 
point being to try it both ways ☺

 

From: Af  on behalf of George Skorup 

Reply-To: 
Date: Sunday, April 8, 2018 at 5:29 PM
To: 
Subject: Re: [AFMUG] OT SIP issue

 

What kind of router/firewall are you working with? No audio is usually a SIP 
ALG thing. You need the ALG on to rewrite the SIP headers when behind NAT.

On 4/8/2018 2:30 PM, Chuck McCown wrote:

Tried both ways, no joy.

Sent from my iPhone


On Apr 8, 2018, at 1:23 PM, Forrest Christian (List Account) 
 wrote:

Are they behind nat? 

 

Sounds like it might be a reinvite issue, asterisk will try to get out of the 
audio path by telling the endpoints to talk directly to each other.   If nat is 
involved asterisk will often tell the endpoints to talk directly even if they 
have no direct connection between them.

 

Disabling reinvite may help if this is the case.

 

On Sun, Apr 8, 2018, 11:27 AM Chuck McCown  wrote:

Pulling our hair out.  The Aastra phones will call each other, we can send dtmf 
but no audio.  Linksys sip ata does the same thing.  These were working, very 
frustrating.  

Re: [AFMUG] OT SIP issue

[George Skorup]

What kind of router/firewall are you working with? No audio is usually a 
SIP ALG thing. You need the ALG on to rewrite the SIP headers when 
behind NAT.


On 4/8/2018 2:30 PM, Chuck McCown wrote:

Tried both ways, no joy.

Sent from my iPhone

On Apr 8, 2018, at 1:23 PM, Forrest Christian (List Account) 
mailto:li...@packetflux.com>> wrote:



Are they behind nat?

Sounds like it might be a reinvite issue, asterisk will try to get 
out of the audio path by telling the endpoints to talk directly to 
each other.   If nat is involved asterisk will often tell the 
endpoints to talk directly even if they have no direct connection 
between them.


Disabling reinvite may help if this is the case.

On Sun, Apr 8, 2018, 11:27 AM Chuck McCown > wrote:


Pulling our hair out.  The Aastra phones will call each other, we
can send dtmf but no audio. Linksys sip ata does the same thing. 
These were working, very frustrating.

Re: [AFMUG] OT SIP issue

[Chuck McCown]

Tried both ways, no joy.

Sent from my iPhone

> On Apr 8, 2018, at 1:23 PM, Forrest Christian (List Account) 
>  wrote:
> 
> Are they behind nat?
> 
> Sounds like it might be a reinvite issue, asterisk will try to get out of the 
> audio path by telling the endpoints to talk directly to each other.   If nat 
> is involved asterisk will often tell the endpoints to talk directly even if 
> they have no direct connection between them.
> 
> Disabling reinvite may help if this is the case.
> 
>> On Sun, Apr 8, 2018, 11:27 AM Chuck McCown  wrote:
>> Pulling our hair out.  The Aastra phones will call each other, we can send 
>> dtmf but no audio.  Linksys sip ata does the same thing.  These were 
>> working, very frustrating. 

Re: [AFMUG] OT SIP issue

[Forrest Christian (List Account)]

Are they behind nat?

Sounds like it might be a reinvite issue, asterisk will try to get out of
the audio path by telling the endpoints to talk directly to each other.
 If nat is involved asterisk will often tell the endpoints to talk directly
even if they have no direct connection between them.

Disabling reinvite may help if this is the case.

On Sun, Apr 8, 2018, 11:27 AM Chuck McCown  wrote:

> Pulling our hair out.  The Aastra phones will call each other, we can send
> dtmf but no audio.  Linksys sip ata does the same thing.  These were
> working, very frustrating.
>

[AFMUG] OT SIP issue

[Chuck McCown]

Pulling our hair out.  The Aastra phones will call each other, we can send dtmf 
but no audio.  Linksys sip ata does the same thing.  These were working, very 
frustrating.