Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] IAX2 bridge failing


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
mdupuis at ocg.ca
Guest





PostPosted: Thu Dec 12, 2013 5:08 pm    Post subject: [asterisk-users] IAX2 bridge failing Reply with quote

I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks.

When I initiate a call from the IAX ATA, something goes wrong. One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's a clue? The dialplan seems to execute properly, and I can watch the destination system which answers the call and starts playing media (monkeys) which I don't hear.

Any ideas on what is going on? Since this is IAX in and IAX out, NAT should not be an issue (even through there is NAT on both sides). Since media moves on the same UDP port as call setup, also proves should not be a network problem (I think)

Can someone point me to a solution?

Thanks!


(IP's and ISP and phone number disguised)

- Executing [s@macro-dialexternal:57] GotoIf("IAX2/S-14468", "1?dialnormal") in new stack
-- Goto (macro-dialexternal,s,60)
-- Executing [s@macro-dialexternal:60] Dial("IAX2/S-14468", "IAX2/ISP123/1234567890|60|W") in new stack
-- Called ISP123/1234567890
-- Call accepted by 201.191.37.138 (format ulaw)
-- Format for call is ulaw
-- IAX2/ISP123-2261 answered IAX2/S-14468
-- Channel 'IAX2/S-14468' ready to transfer
-- Channel 'IAX2/ISP123-2261' ready to transfer
-- Hungup 'IAX2/ISP123-2261'
Back to top
mdupuis at ocg.ca
Guest





PostPosted: Fri Dec 13, 2013 11:42 am    Post subject: [asterisk-users] IAX2 bridge failing Reply with quote

Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero.

While the channels are up, I did an core show channel xxx and found Blocking in:
ast_waitfor_nandfds

Is this a bug? Or something I can fix through config?

From: asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Michelle Dupuis [mdupuis@ocg.ca]
Sent: Thursday, December 12, 2013 5:08 PM
To: Asterisk Users List
Subject: [asterisk-users] IAX2 bridge failing



I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks.

When I initiate a call from the IAX ATA, something goes wrong. One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's a clue? The dialplan seems to execute properly, and I can watch the destination system which answers the call and starts playing media (monkeys) which I don't hear.

Any ideas on what is going on? Since this is IAX in and IAX out, NAT should not be an issue (even through there is NAT on both sides). Since media moves on the same UDP port as call setup, also proves should not be a network problem (I think)

Can someone point me to a solution?

Thanks!


(IP's and ISP and phone number disguised)

- Executing [s@macro-dialexternal:57] GotoIf("IAX2/S-14468", "1?dialnormal") in new stack
-- Goto (macro-dialexternal,s,60)
-- Executing [s@macro-dialexternal:60] Dial("IAX2/S-14468", "IAX2/ISP123/1234567890|60|W") in new stack
-- Called ISP123/1234567890
-- Call accepted by 201.191.37.138 (format ulaw)
-- Format for call is ulaw
-- IAX2/ISP123-2261 answered IAX2/S-14468
-- Channel 'IAX2/S-14468' ready to transfer
-- Channel 'IAX2/ISP123-2261' ready to transfer
-- Hungup 'IAX2/ISP123-2261'
Back to top
jcolp at digium.com
Guest





PostPosted: Fri Dec 13, 2013 11:44 am    Post subject: [asterisk-users] IAX2 bridge failing Reply with quote

Michelle Dupuis wrote:
Quote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx and found
Blocking in:
ast_waitfor_nandfds
Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the
peers/users/friends/etc in question, reload, retry, and see if that
changes the behavior. If it does then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
mdupuis at ocg.ca
Guest





PostPosted: Fri Dec 13, 2013 11:48 am    Post subject: [asterisk-users] IAX2 bridge failing Reply with quote

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" inbetween)....same result

I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine).

Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x)
________________________________________
From: asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Joshua Colp [jcolp@digium.com]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
Quote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx and found
Blocking in:
ast_waitfor_nandfds
Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the
peers/users/friends/etc in question, reload, retry, and see if that
changes the behavior. If it does then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
mdupuis at ocg.ca
Guest





PostPosted: Sat Dec 14, 2013 11:21 pm    Post subject: [asterisk-users] IAX2 bridge failing Reply with quote

Ok just restart

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" inbetween)....same result

I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine).

Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) ________________________________________
From: asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Joshua Colp [jcolp@digium.com]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
Quote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx and found
Blocking in:
ast_waitfor_nandfds
Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
mdupuis at ocg.ca
Guest





PostPosted: Sat Dec 14, 2013 11:27 pm    Post subject: [asterisk-users] IAX2 bridge failing Reply with quote

meant to say restart didn't help either..

________________________________________
From: asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Michelle Dupuis [mdupuis@ocg.ca]
Sent: Saturday, December 14, 2013 11:20 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Ok just restart

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" inbetween)....same result

I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine).

Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) ________________________________________
From: asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Joshua Colp [jcolp@digium.com]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
Quote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx and found
Blocking in:
ast_waitfor_nandfds
Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
stdavis at multiservic...
Guest





PostPosted: Sun Dec 15, 2013 12:41 am    Post subject: [asterisk-users] IAX2 bridge failing Reply with quote

Did you change your network switch recently?  Some Digium IAX ATAs do not behave well with Cisco equipment.


On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis <mdupuis@ocg.ca (mdupuis@ocg.ca)> wrote:
Quote:
meant to say restart didn't help either..

________________________________________
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Michelle Dupuis [mdupuis@ocg.ca (mdupuis@ocg.ca)]
Sent: Saturday, December 14, 2013 11:20 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Ok just restart

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" inbetween)....same result

I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine).

Anything else to suggest?  I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) ________________________________________
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Joshua Colp [jcolp@digium.com (jcolp@digium.com)]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
Quote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx and found
Blocking in:
ast_waitfor_nandfds
Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:  www.digium.com  & www.asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users






--
Steven DavisVoIP Engineer
Multi Service 

+1-913-663-9748 o
+1-913-871-5155 m

stdavis@multiservice.com (stdavis@multiservice.com)

[url=http://www.multiservice.com/][/url] 




Quote:
------------------------------------------------------------------
This email is intended solely for the use of the addressee and may
contain information that is confidential, proprietary, or both.
If you receive this email in error please immediately notify the
sender and delete the email..
------------------------------------------------------------------
Back to top
mdupuis at ocg.ca
Guest





PostPosted: Sun Dec 15, 2013 8:12 am    Post subject: [asterisk-users] IAX2 bridge failing Reply with quote

No - but this is a new setup so I can't say it worked before...it just isn't working from the start.

I've found the call setup works and once bridged there is one way audio (to the ATA, none from the ATA). And the the connection drops after 30 secs approx because something on the path (or endpoint) realizes something is wrong...

From: asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Steven Davis [stdavis@multiservice.com]
Sent: Sunday, December 15, 2013 12:41 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing



Did you change your network switch recently? Some Digium IAX ATAs do not behave well with Cisco equipment.


On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis <mdupuis@ocg.ca (mdupuis@ocg.ca)> wrote:
Quote:
meant to say restart didn't help either..

________________________________________
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Michelle Dupuis [mdupuis@ocg.ca (mdupuis@ocg.ca)]
Sent: Saturday, December 14, 2013 11:20 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Ok just restart

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" inbetween)....same result

I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine).

Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) ________________________________________
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Joshua Colp [jcolp@digium.com (jcolp@digium.com)]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
Quote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx and found
Blocking in:
ast_waitfor_nandfds
Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users






--
Steven Davis VoIP Engineer
Multi Service

+1-913-663-9748 o
+1-913-871-5155 m

stdavis@multiservice.com (stdavis@multiservice.com)

[url=http://www.multiservice.com/][/url]




Quote:
------------------------------------------------------------------
This email is intended solely for the use of the addressee and may
contain information that is confidential, proprietary, or both.
If you receive this email in error please immediately notify the
sender and delete the email..
------------------------------------------------------------------
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services