[asterisk-users] chan_ss7 with ringing, but no voice stream.
hello, all of users:
sorry, resend it again for clarifying the message. I have implemented cha_ss7
in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think the ss7 does not send the voice steam to the destination.
in chan_ss7, i added:
===
static struct ss7_chan *cic_hunt_even_mru(struct linkset*
linkset) {
struct ss7_chan *cur, *prev, *best, *best_prev;
best = NULL;
best_prev = NULL;
for(cur = linkset-idle_list, prev = NULL; cur !=
NULL; prev = cur, cur = cur-next_idle) {
/* Don't select lines that are resetting or
blocked. */
if(!cur-reset_done || (cur-blocked
(BL_LH|BL_RM|BL_RH|BL_UNEQUIPPED|BL_LINKDOWN))) {
continue;
}
/* if((cur-cic % 2) == 0) { */
/*change to this*/
if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
/* Choose the first idle even circuit,
if any. */
/*end of change*/
best = cur;
best_prev = prev;
break;
} else if(best == NULL) {
/* Remember the first odd circuit, in
case no even circuits are
available. */
best = cur;
best_prev = prev;
}
}
cic_hunt_even_mru if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
my environment is:
asterisk-1.4.20
chan_ss7-1.0.91
Openvox D410P
===
anyone has an idea for the problem?
please give me some hints!
thanks!
james.zhu
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Re: [asterisk-users] chan_ss7 with ringing, but no voice stream.
SS7 doesnt send any voice. It sends call info, and tells the switches
which trunk to use for the voice. Trunks are two-way as far as audio
content, though they maybe designated is inbound or outbound trunks.
An audio problem is possibly a NAT or other issue.
Since you are modifying the SS7 code, there could be some error in setting
up the call, but normally the IMT trunks are two way. (Of course they are 4
wire circuits so are two one way paths, but they are matched pairs so,
for practical purposes they would be 1 entity for call set up purposes.)
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of lizhong zhu
Sent: Friday, March 20, 2009 2:05 AM
To: asterisk-ss7
Subject: [asterisk-users] chan_ss7 with ringing, but no voice stream.
hello, all of users:
sorry, resend it again for clarifying the message. I have implemented
cha_ss7 in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think the ss7 does not send the voice steam to the destination.
in chan_ss7, i added:
===
static struct ss7_chan *cic_hunt_even_mru(struct linkset*
linkset) {
struct ss7_chan *cur, *prev, *best, *best_prev;
best = NULL;
best_prev = NULL;
for(cur = linkset-idle_list, prev = NULL; cur !=
NULL; prev = cur, cur = cur-next_idle) {
/* Don't select lines that are resetting or
blocked. */
if(!cur-reset_done || (cur-blocked
(BL_LH|BL_RM|BL_RH|BL_UNEQUIPPED|BL_LINKDOWN))) {
continue;
}
/* if((cur-cic % 2) == 0) { */
/*change to this*/
if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
/* Choose the first idle even circuit,
if any. */
/*end of change*/
best = cur;
best_prev = prev;
break;
} else if(best == NULL) {
/* Remember the first odd circuit, in
case no even circuits are
available. */
best = cur;
best_prev = prev;
}
}
cic_hunt_even_mru if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
my environment is:
asterisk-1.4.20
chan_ss7-1.0.91
Openvox D410P
===
anyone has an idea for the problem?
please give me some hints!
thanks!
james.zhu
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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] chan_ss7 with ringing, but no voice stream.
Cary Fitch wrote:
SS7 doesn’t send any voice. It sends call info, and tells the switches
which trunk to use for the voice. Trunks are two-way as far as audio
content, though they maybe designated is inbound or outbound trunks.
An audio problem is possibly a NAT or other issue.
Since you are modifying the SS7 code, there could be some error in setting
up the call, but normally the IMT trunks are two way. (Of course they are 4
wire circuits so are two one way paths, but they are matched pairs so,
for practical purposes they would be 1 entity for call set up purposes.)
Actually, the implementations of SS7 support in Asterisk (libss7, and
also the out of tree chan_ss7) include support for signaling and bearer
channels, which is why he's mentioning voice support.
Right now, both implementations function basically like the ISDN code
works - i.e. you have to terminate signaling and bearer channels on the
same box.
Matthew Fredrickson (the libss7 guy :-) )
Digium, Inc.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of lizhong zhu
Sent: Friday, March 20, 2009 2:05 AM
To: asterisk-ss7
Subject: [asterisk-users] chan_ss7 with ringing, but no voice stream.
hello, all of users:
sorry, resend it again for clarifying the message. I have implemented
cha_ss7 in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think the ss7 does not send the voice steam to the destination.
in chan_ss7, i added:
===
static struct ss7_chan *cic_hunt_even_mru(struct linkset*
linkset) {
struct ss7_chan *cur, *prev, *best, *best_prev;
best = NULL;
best_prev = NULL;
for(cur = linkset-idle_list, prev = NULL; cur !=
NULL; prev = cur, cur = cur-next_idle) {
/* Don't select lines that are resetting or
blocked. */
if(!cur-reset_done || (cur-blocked
(BL_LH|BL_RM|BL_RH|BL_UNEQUIPPED|BL_LINKDOWN))) {
continue;
}
/* if((cur-cic % 2) == 0) { */
/*change to this*/
if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
/* Choose the first idle even circuit,
if any. */
/*end of change*/
best = cur;
best_prev = prev;
break;
} else if(best == NULL) {
/* Remember the first odd circuit, in
case no even circuits are
available. */
best = cur;
best_prev = prev;
}
}
cic_hunt_even_mru if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
my environment is:
asterisk-1.4.20
chan_ss7-1.0.91
Openvox D410P
===
anyone has an idea for the problem?
please give me some hints!
thanks!
james.zhu
___
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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