hi all
configuration:
inalp gw (pstn) ------> ser ------> asterisk (only voicemail)
the following problems occur:
1) ser forwards the call to asterisk (if the user is in the voicemail group) if the client does not answer after 30 sec. from sip to sip is working. but if a call comes from the pstn to a sip client, the client rings and hangs up after 30 sec. ser then forwards it to the asterisk, but the pstn call keeps on ringing and no voicemail is to hear.
ser writes the following error
Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++INVITE****************** Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:20 sip /usr/local/sbin/ser[7017]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:21 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:24 sip last message repeated 3 times Mar 29 13:25:28 sip /usr/local/sbin/ser[7015]: ++++++++++++Loose Route****************** Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: contact_parser(): Empty body Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: parse_contact(): Error while parsing Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: get_contact_uri: Error while parsing Contact body
2) if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
does anyone know about these problems?
regards raid
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 25) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (lookup("location") | lookup("aliases")) { if (is_user_in("Request-URI", "voicemail")) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); t_relay(); route(2); break; } else { log (1, "++++++++++++no voicemail********************"); route(2); break; } }
}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehostport("XXX.XXX.XXX.XXX:5060"); append_branch(); t_on_failure("1"); t_relay(); break();
}
is the call hitting asterisk, if so is asterisk calling the voicemail file
hofi wrote:
hi all
configuration:
inalp gw (pstn) ------> ser ------> asterisk (only voicemail)
the following problems occur:
- ser forwards the call to asterisk (if the user is in the voicemail
group) if the client does not answer after 30 sec. from sip to sip is working. but if a call comes from the pstn to a sip client, the client rings and hangs up after 30 sec. ser then forwards it to the asterisk, but the pstn call keeps on ringing and no voicemail is to hear.
ser writes the following error
Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++INVITE****************** Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:20 sip /usr/local/sbin/ser[7017]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:21 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:24 sip last message repeated 3 times Mar 29 13:25:28 sip /usr/local/sbin/ser[7015]: ++++++++++++Loose Route****************** Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: contact_parser(): Empty body Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: parse_contact(): Error while parsing Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: get_contact_uri: Error while parsing Contact body
- if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up
ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
does anyone know about these problems?
regards raid
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 25) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (lookup("location") | lookup("aliases")) { if (is_user_in("Request-URI", "voicemail")) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); t_relay(); route(2); break; } else { log (1, "++++++++++++no voicemail********************"); route(2); break; } }
}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehostport("XXX.XXX.XXX.XXX:5060"); append_branch(); t_on_failure("1"); t_relay(); break();
}
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
.
sorry
ser log sip to sip:
Mar 29 22:05:30 sip /usr/local/sbin/ser[8025]: ++++++++++++INVITE****************** Mar 29 22:05:30 sip /usr/local/sbin/ser[8025]: ++++++++++++voicemail*********************** Mar 29 22:06:10 sip /usr/local/sbin/ser[8030]: ++++++++++++Failure Route****************** Mar 29 22:06:10 sip /usr/local/sbin/ser[8024]: ++++++++++++Loose Route****************** Mar 29 22:06:19 sip /usr/local/sbin/ser[8025]: ++++++++++++INVITE****************** Mar 29 22:06:19 sip /usr/local/sbin/ser[8025]: ++++++++++++Loose Route******************
and asterisk log sip to sip:
asterisk*CLI> -- Executing Answer("SIP/sip.xxx.xxx-0810e610", "") in new stack -- Executing VoiceMail("SIP/sip.xxx.xxx-0810e610", "u798622211") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/9' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/6' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Executing Answer("SIP/sip.xxx.xxx-0810e610", "") in new stack
ser log pstn to sip:
Mar 29 22:10:11 sip /usr/local/sbin/ser[8024]: ++++++++++++INVITE****************** Mar 29 22:10:11 sip /usr/local/sbin/ser[8024]: ++++++++++++voicemail*********************** Mar 29 22:10:11 sip /usr/local/sbin/ser[8024]: ++++++++++++INVITE****************** Mar 29 22:10:11 sip /usr/local/sbin/ser[8024]: ++++++++++++voicemail*********************** Mar 29 22:10:52 sip /usr/local/sbin/ser[8030]: ++++++++++++Failure Route****************** Mar 29 22:10:52 sip /usr/local/sbin/ser[8030]: ++++++++++++Failure Route****************** Mar 29 22:10:52 sip /usr/local/sbin/ser[8024]: ++++++++++++voicemail*********************** Mar 29 22:10:52 sip /usr/local/sbin/ser[8022]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 22:10:53 sip /usr/local/sbin/ser[8024]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 22:10:57 sip last message repeated 4 times Mar 29 22:11:02 sip /usr/local/sbin/ser[8022]: ++++++++++++Loose Route****************** Mar 29 22:11:02 sip /usr/local/sbin/ser[8024]: ++++++++++++voicemail*********************** Mar 29 22:11:04 sip last message repeated 2 times Mar 29 22:11:05 sip /usr/local/sbin/ser[8023]: ++++++++++++voicemail*********************** Mar 29 22:11:07 sip last message repeated 2 times
asterisk log pstn to sip:
asterisk*CLI> -- Executing Answer("SIP/sip.xxx.xxx-0810e610", "") in new stack -- Executing VoiceMail("SIP/sip.xxx.xxx-0810e610", "u798622211") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/9' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/6' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/2' (language 'en') Mar 29 22:02:07 WARNING[1160]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 90f4fda6fc5a64089c7077e0024b6199@sip.xxx.xxx for seqno 529261175 (Non-critical Response) -- Playing 'digits/1' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') Mar 29 22:02:15 WARNING[1588]: file.c:550 ast_readaudio_callback: Failed to write frame -- Executing Answer("SIP/sip.xxx.xxx-0810e610", "") in new stack Mar 29 22:02:21 WARNING[1160]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 90f4fda6fc5a64089c7077e0024b6199@sip.xxx.xxx for seqno 529261175 (Critical Response)
regards hofi
is the call hitting asterisk, if so is asterisk calling the voicemail file
hofi wrote:
hi all
configuration:
inalp gw (pstn) ------> ser ------> asterisk (only voicemail)
the following problems occur:
- ser forwards the call to asterisk (if the user is in the voicemail
group) if the client does not answer after 30 sec. from sip to sip is working. but if a call comes from the pstn to a sip client, the client rings and hangs up after 30 sec. ser then forwards it to the asterisk, but the pstn call keeps on ringing and no voicemail is to hear.
ser writes the following error
Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++INVITE****************** Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:20 sip /usr/local/sbin/ser[7017]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:21 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:24 sip last message repeated 3 times Mar 29 13:25:28 sip /usr/local/sbin/ser[7015]: ++++++++++++Loose Route****************** Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: contact_parser(): Empty body Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: parse_contact(): Error while parsing Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: get_contact_uri: Error while parsing Contact body
- if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up
ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
does anyone know about these problems?
regards raid
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 25) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (lookup("location") | lookup("aliases")) { if (is_user_in("Request-URI", "voicemail")) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); t_relay(); route(2); break; } else { log (1, "++++++++++++no voicemail********************"); route(2); break; } }
}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehostport("XXX.XXX.XXX.XXX:5060"); append_branch(); t_on_failure("1"); t_relay(); break();
}
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
.
I never did figure out exactly what the cause was [sorry], but we were looking into a secondary carrier and ran into the same problems you describe below. We took one of our Cisco AS5350's to their NOC to put in place of their nextone and everything worked just fine. You may want to look at the settings on your inalp. We were unable to get it to work with the nextone. My guess was that it had something to do with codec translations but I have no data to back that up. All the SIP messaging seemed to be there.
I'd be interested in what fixes this, so if you discover a solution, please post to the list.
Thanks.
dan
On Tue, 29 Mar 2005 20:36:08 +0200 (CEST), hofi raid@hofi.priv.at wrote:
hi all
configuration:
inalp gw (pstn) ------> ser ------> asterisk (only voicemail)
the following problems occur:
- ser forwards the call to asterisk (if the user is in the voicemail
group) if the client does not answer after 30 sec. from sip to sip is working. but if a call comes from the pstn to a sip client, the client rings and hangs up after 30 sec. ser then forwards it to the asterisk, but the pstn call keeps on ringing and no voicemail is to hear.
ser writes the following error
Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++INVITE****************** Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:20 sip /usr/local/sbin/ser[7017]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:21 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:24 sip last message repeated 3 times Mar 29 13:25:28 sip /usr/local/sbin/ser[7015]: ++++++++++++Loose Route****************** Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: contact_parser(): Empty body Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: parse_contact(): Error while parsing Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: get_contact_uri: Error while parsing Contact body
- if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up
ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
does anyone know about these problems?
regards raid
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 25) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (lookup("location") | lookup("aliases")) { if (is_user_in("Request-URI", "voicemail")) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); t_relay(); route(2); break; } else { log (1, "++++++++++++no voicemail********************"); route(2); break; } }
}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehostport("XXX.XXX.XXX.XXX:5060"); append_branch(); t_on_failure("1"); t_relay(); break();
}
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
hi
the first problem is solved (pstn to voicemail !working!)
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 40) modparam("tm", "fr_inv_timer", 35) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (is_user_in("Request-URI", "voicemail")) { setflag(4); };
if (lookup("location") | lookup("aliases")) { if (method == "INVITE" && isflagset(4)) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); } route(2); break; };
}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehost("XXX.XXX.XXX.XXX"); append_branch(); t_relay_to_udp("XXX.XXX.XXX.XXX", "5060"); }
but for my second problem I do not find the error
2) if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
thx
regards hofi
I never did figure out exactly what the cause was [sorry], but we were looking into a secondary carrier and ran into the same problems you describe below. We took one of our Cisco AS5350's to their NOC to put in place of their nextone and everything worked just fine. You may want to look at the settings on your inalp. We were unable to get it to work with the nextone. My guess was that it had something to do with codec translations but I have no data to back that up. All the SIP messaging seemed to be there.
I'd be interested in what fixes this, so if you discover a solution, please post to the list.
Thanks.
dan
On Tue, 29 Mar 2005 20:36:08 +0200 (CEST), hofi raid@hofi.priv.at wrote:
hi all
configuration:
inalp gw (pstn) ------> ser ------> asterisk (only voicemail)
the following problems occur:
- ser forwards the call to asterisk (if the user is in the voicemail
group) if the client does not answer after 30 sec. from sip to sip is working. but if a call comes from the pstn to a sip client, the client rings and hangs up after 30 sec. ser then forwards it to the asterisk, but the pstn call keeps on ringing and no voicemail is to hear.
ser writes the following error
Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++INVITE****************** Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:20 sip /usr/local/sbin/ser[7017]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:21 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:24 sip last message repeated 3 times Mar 29 13:25:28 sip /usr/local/sbin/ser[7015]: ++++++++++++Loose Route****************** Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: contact_parser(): Empty body Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: parse_contact(): Error while parsing Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: get_contact_uri: Error while parsing Contact body
- if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up
ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
does anyone know about these problems?
regards raid
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 25) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (lookup("location") | lookup("aliases")) { if (is_user_in("Request-URI", "voicemail")) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); t_relay(); route(2); break; } else { log (1, "++++++++++++no voicemail********************"); route(2); break; } }
}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehostport("XXX.XXX.XXX.XXX:5060"); append_branch(); t_on_failure("1"); t_relay(); break();
}
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi Hofi You must use t_check_status... this will solve your problem:
failure_route[1]{ if (t_check_status("408|486")){ ## Do not include 487!! revert_uri (); rewritehostport ("x.x.x.x:port"); append_branch(); t_relay(); break; } }
regards
Rafael J. Risco G.V. Millicom Peru SA
On Wed, 30 Mar 2005 11:58:32 +0200 (CEST), hofi raid@hofi.priv.at wrote:
hi
the first problem is solved (pstn to voicemail !working!)
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 40) modparam("tm", "fr_inv_timer", 35) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (is_user_in("Request-URI", "voicemail")) { setflag(4); };
if (lookup("location") | lookup("aliases")) { if (method == "INVITE" && isflagset(4)) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); } route(2); break; };}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehost("XXX.XXX.XXX.XXX"); append_branch(); t_relay_to_udp("XXX.XXX.XXX.XXX", "5060"); }
but for my second problem I do not find the error
- if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up
ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
thx
regards hofi
I never did figure out exactly what the cause was [sorry], but we were looking into a secondary carrier and ran into the same problems you describe below. We took one of our Cisco AS5350's to their NOC to put in place of their nextone and everything worked just fine. You may want to look at the settings on your inalp. We were unable to get it to work with the nextone. My guess was that it had something to do with codec translations but I have no data to back that up. All the SIP messaging seemed to be there.
I'd be interested in what fixes this, so if you discover a solution, please post to the list.
Thanks.
dan
On Tue, 29 Mar 2005 20:36:08 +0200 (CEST), hofi raid@hofi.priv.at wrote:
hi all
configuration:
inalp gw (pstn) ------> ser ------> asterisk (only voicemail)
the following problems occur:
- ser forwards the call to asterisk (if the user is in the voicemail
group) if the client does not answer after 30 sec. from sip to sip is working. but if a call comes from the pstn to a sip client, the client rings and hangs up after 30 sec. ser then forwards it to the asterisk, but the pstn call keeps on ringing and no voicemail is to hear.
ser writes the following error
Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++INVITE****************** Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:20 sip /usr/local/sbin/ser[7017]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:21 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Mar 29 13:25:24 sip last message repeated 3 times Mar 29 13:25:28 sip /usr/local/sbin/ser[7015]: ++++++++++++Loose Route****************** Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: contact_parser(): Empty body Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: parse_contact(): Error while parsing Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: get_contact_uri: Error while parsing Contact body
- if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up
ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
does anyone know about these problems?
regards raid
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 25) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (lookup("location") | lookup("aliases")) { if (is_user_in("Request-URI", "voicemail")) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); t_relay(); route(2); break; } else { log (1, "++++++++++++no voicemail********************"); route(2); break; } }
}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehostport("XXX.XXX.XXX.XXX:5060"); append_branch(); t_on_failure("1"); t_relay(); break();
}
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
hi rafael
yes that's it :)
thank you very much
regards hofi
Hi Hofi You must use t_check_status... this will solve your problem:
failure_route[1]{ if (t_check_status("408|486")){ ## Do not include 487!! revert_uri (); rewritehostport ("x.x.x.x:port"); append_branch(); t_relay(); break; } }
regards
Rafael J. Risco G.V. Millicom Peru SA
On Wed, 30 Mar 2005 11:58:32 +0200 (CEST), hofi raid@hofi.priv.at wrote:
hi
the first problem is solved (pstn to voicemail !working!)
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 40) modparam("tm", "fr_inv_timer", 35) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (is_user_in("Request-URI", "voicemail")) { setflag(4); };
if (lookup("location") | lookup("aliases")) { if (method == "INVITE" && isflagset(4)) { log (1, "++++++++++++voicemail***********************"); t_on_failure("1"); avp_write("i:30", "inv_timeout"); } route(2); break; };}
route [2] { if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehost("XXX.XXX.XXX.XXX"); append_branch(); t_relay_to_udp("XXX.XXX.XXX.XXX", "5060"); }
but for my second problem I do not find the error
- if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up
ser forwards to asterisk and the voicemail is run in the background, but instead the call should be cancelled and not forwarded to the voicemail
thx
regards hofi
I never did figure out exactly what the cause was [sorry], but we were looking into a secondary carrier and ran into the same problems you describe below. We took one of our Cisco AS5350's to their NOC to put in place of their nextone and everything worked just fine. You may want to look at the settings on your inalp. We were unable to get it to work with the nextone. My guess was that it had something to do with codec translations but I have no data to back that up. All the SIP messaging seemed to be there.
I'd be interested in what fixes this, so if you discover a solution, please post to the list.
Thanks.
dan
On Tue, 29 Mar 2005 20:36:08 +0200 (CEST), hofi raid@hofi.priv.at
wrote:
hi all
configuration:
inalp gw (pstn) ------> ser ------> asterisk (only voicemail)
the following problems occur:
- ser forwards the call to asterisk (if the user is in the voicemail
group) if the client does not answer after 30 sec. from sip to sip is working. but if a call comes from the pstn to a sip client, the client rings
and
hangs up after 30 sec. ser then forwards it to the asterisk, but the pstn call keeps on ringing and no voicemail is to hear.
ser writes the following error
Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++INVITE****************** Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]: ++++++++++++Failure Route****************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7017]: ++++++++++++voicemail*********************** Mar 29 13:25:19 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408,
received:
200 Mar 29 13:25:20 sip /usr/local/sbin/ser[7017]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408,
received:
200 Mar 29 13:25:21 sip /usr/local/sbin/ser[7016]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408,
received:
200 Mar 29 13:25:24 sip last message repeated 3 times Mar 29 13:25:28 sip /usr/local/sbin/ser[7015]: ++++++++++++Loose Route****************** Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: contact_parser():
Empty
body Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: parse_contact(): Error while parsing Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: get_contact_uri: Error while parsing Contact body
- if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs
up
ser forwards to asterisk and the voicemail is run in the background,
but
instead the call should be cancelled and not forwarded to the
voicemail
does anyone know about these problems?
regards raid
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 25) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route { if (lookup("location") | lookup("aliases")) { if (is_user_in("Request-URI", "voicemail")) { log (1,
"++++++++++++voicemail***********************");
t_on_failure("1"); avp_write("i:30", "inv_timeout"); t_relay(); route(2); break; } else { log (1, "++++++++++++no
voicemail********************");
route(2); break; } }
}
route [2] { if
(uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)"
&& !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(2)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; }
failure_route[1] { log (1, "++++++++++++Failure Route******************"); revert_uri(); rewritehostport("XXX.XXX.XXX.XXX:5060"); append_branch(); t_on_failure("1"); t_relay(); break();
}
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
--
rrgv
-
Daniel Poulsen -
hofi -
Iqbal -
Rafael J. Risco G.V.