[Serusers] ser+asterisk+voicemail

hofi raid at hofi.priv.at
Tue Mar 29 21:57:10 CEST 2005 

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 at 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 at 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:
>>
>>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();
>>
>>}
>>
>>
>>
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers at lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>>.
>>
>>
>>
>

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