Hello,
I am having a problem with transferring calls that come in from the
outside. Users have been calling in to the PRI that's on the Cisco GW,
then they are passed into Asterisk via SIP and to the end phone (Polycom
501/601) using SIP. When that user tries to transfer that call to
another extension, the call disconnects and hangs in the air and doesn't
do anything. The call shows active in the Cisco GW but no where to be
found in asterisk. Here is some log output of a transfer attempt:
-- Stopped music on hold on SIP/10.25.118.2-b7b4e520
== Spawn extension (ANC, 4023, 2) exited non-zero on
'SIP/4023-ebbf<ZOMBIE>'
-- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520
-- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and
SIP/2198-3780
-- Incoming call: Got SIP response 500 "Internal Server Error"
back
from 10.45.25.12
I'm not sure if the SIP 500 error is relative to my issue. Any ideas on
what could be causing SIP transfers to hang or drop?
Thank you,
Dan
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Asterisk 1.2.9.1 is the version I'm on.
_____
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dan
Brummer
Sent: Tuesday, July 11, 2006 8:30 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Issues with making Transfers
Hello,
I am having a problem with transferring calls that come in from the
outside. Users have been calling in to the PRI that's on the Cisco GW,
then they are passed into Asterisk via SIP and to the end phone (Polycom
501/601) using SIP. When that user tries to transfer that call to
another extension, the call disconnects and hangs in the air and doesn't
do anything. The call shows active in the Cisco GW but no where to be
found in asterisk. Here is some log output of a transfer attempt:
-- Stopped music on hold on SIP/10.25.118.2-b7b4e520
== Spawn extension (ANC, 4023, 2) exited non-zero on
'SIP/4023-ebbf<ZOMBIE>'
-- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520
-- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and
SIP/2198-3780
-- Incoming call: Got SIP response 500 "Internal Server Error"
back
from 10.45.25.12
I'm not sure if the SIP 500 error is relative to my issue. Any ideas on
what could be causing SIP transfers to hang or drop?
Thank you,
Dan
-------------- next part --------------
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Thank you for the response, I will try downgrading. -Dan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ron Wellsted Sent: Tuesday, July 11, 2006 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issues with making Transfers -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dan Brummer wrote:> Asterisk 1.2.9.1 is the version I'm on. > > ---------------------------------------------------------------------- > -- > *From:* asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] *On Behalf Of *Dan > Brummer > *Sent:* Tuesday, July 11, 2006 8:30 AM > *To:* asterisk-users@lists.digium.com > *Subject:* [asterisk-users] Issues with making Transfers > > Hello, > I am having a problem with transferring calls that come in from the > outside. Users have been calling in to the PRI that's on the Cisco > GW, then they are passed into Asterisk via SIP and to the end phone > (Polycom > 501/601) using SIP. When that user tries to transfer that call to > another extension, the call disconnects and hangs in the air and > doesn't do anything. The call shows active in the Cisco GW but no > where to be found in asterisk. Here is some log output of a transferattempt:> > -- Stopped music on hold on SIP/10.25.118.2-b7b4e520 > == Spawn extension (ANC, 4023, 2) exited non-zero on > 'SIP/4023-ebbf<ZOMBIE>' > -- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520 > -- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and > SIP/2198-3780 > -- Incoming call: Got SIP response 500 "Internal Server Error" > back from 10.45.25.12 > > > I'm not sure if the SIP 500 error is relative to my issue. Any ideas > on what could be causing SIP transfers to hang or drop? > > Thank you, > DanInterestingly, we saw a very similar issue with 1.2.9.1 and Cisco 7960s (SIP 8.2 fw) and HFC BRI ISDN cards last week, I went back to 1.2.7 and every thing seems fine now. - -- Ron Wellsted ron@wellsted.org.uk http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRLP1n0tP/KMNOfRbAQI86gf7BW1G8CmMzOo3O3Wu200gFGlYEUwbc+8Q tk39rot1H6Bus0O0qPNoSAgJyxWp5617urprU9th2hreRjh5r3Cb3MOIfDuhCm2W p7b1UyVhFZaehWy8ketykld1mvV5eCBBCu9aKYINRS4aEAx7Snt3txLEB5x1bA7A 7N97O/h821iqR79fTuhBD8GMOF0dwaVmJ8oAeeUZoR+YXngMGt2pXQCL7LyLMmT2 vNgusL28J4Cmw76sHuuXEQ8W/t1ONT7WPWWwj/TNFzIqGvl4dCeMC5yN4XtHnXp0 l7gYY9qoFxjD4Z6sXzqiETqFsuqsZygDhsqdMg5CaaUEbi94uERMFQ==0k5S -----END PGP SIGNATURE----- _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
This has worked. I downgraded from 1.2.9.1 to 1.2.7.1 and I'm not having the warm transfer issue anymore. Does anyone know if this is a known issue and is going to be fixed in upcoming release? Should I possibly put in a bug request? -Dan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ron Wellsted Sent: Tuesday, July 11, 2006 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issues with making Transfers -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dan Brummer wrote:> Asterisk 1.2.9.1 is the version I'm on. > > ---------------------------------------------------------------------- > -- > *From:* asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] *On Behalf Of *Dan > Brummer > *Sent:* Tuesday, July 11, 2006 8:30 AM > *To:* asterisk-users@lists.digium.com > *Subject:* [asterisk-users] Issues with making Transfers > > Hello, > I am having a problem with transferring calls that come in from the > outside. Users have been calling in to the PRI that's on the Cisco > GW, then they are passed into Asterisk via SIP and to the end phone > (Polycom > 501/601) using SIP. When that user tries to transfer that call to > another extension, the call disconnects and hangs in the air and > doesn't do anything. The call shows active in the Cisco GW but no > where to be found in asterisk. Here is some log output of a transferattempt:> > -- Stopped music on hold on SIP/10.25.118.2-b7b4e520 > == Spawn extension (ANC, 4023, 2) exited non-zero on > 'SIP/4023-ebbf<ZOMBIE>' > -- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520 > -- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and > SIP/2198-3780 > -- Incoming call: Got SIP response 500 "Internal Server Error" > back from 10.45.25.12 > > > I'm not sure if the SIP 500 error is relative to my issue. Any ideas > on what could be causing SIP transfers to hang or drop? > > Thank you, > DanInterestingly, we saw a very similar issue with 1.2.9.1 and Cisco 7960s (SIP 8.2 fw) and HFC BRI ISDN cards last week, I went back to 1.2.7 and every thing seems fine now. - -- Ron Wellsted ron@wellsted.org.uk http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRLP1n0tP/KMNOfRbAQI86gf7BW1G8CmMzOo3O3Wu200gFGlYEUwbc+8Q tk39rot1H6Bus0O0qPNoSAgJyxWp5617urprU9th2hreRjh5r3Cb3MOIfDuhCm2W p7b1UyVhFZaehWy8ketykld1mvV5eCBBCu9aKYINRS4aEAx7Snt3txLEB5x1bA7A 7N97O/h821iqR79fTuhBD8GMOF0dwaVmJ8oAeeUZoR+YXngMGt2pXQCL7LyLMmT2 vNgusL28J4Cmw76sHuuXEQ8W/t1ONT7WPWWwj/TNFzIqGvl4dCeMC5yN4XtHnXp0 l7gYY9qoFxjD4Z6sXzqiETqFsuqsZygDhsqdMg5CaaUEbi94uERMFQ==0k5S -----END PGP SIGNATURE----- _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users