Hi,

I'll be happy if there is config parameter which can change how many times INVITE request re-sent.

I've had this same problem, trying to do forwarding to voicemail. If the voicemail does not answer (using a separate vm server -- asterisk) then the invite keeps going forever. In fact, the sip phone rings forever even after the original inviter has canceled.

I didn't add the append_branch("sip:parallel@localhost:9"). Is that necessary for some reason? /a

• zaki - wrote:

...

Hi,

Thank you for your reply. I've already used fr_invite_timer. In this case, if SER send INVITE request, but nothing replied by endpoint. So SER try to re-send INVITE request again and again, and I'd like to change the number of retry of INVITE request. Do you have any workaround for this?

Best Regards,

...

You can use the fr_invite_timer:

http://www.iptel.org/ser/doc/seruser/seruser.html#REPLYPROCESSINGSECTION

Klaus

...

-----Original Message----- From: - zaki - [mailto:zaki@edge.jp] Sent: Tuesday, December 09, 2003 6:11 PM To: serusers@lists.iptel.org Subject: [Serusers] stop rety

Hi,

With following situation,

• endpoint exists in location

(found by serctl ul show)

• but endpoint is actually dead

if someone call the endpoint, then SER try to send INVITE request again and again.

Could anyone please let me know how to change the number of retry in this situation?

Best Regards,

--

• zaki - YAMAZAKI Noriyuki, Director of EDGE Co., Ltd. +81-3-5749-9621(Phone), +81-3-5749-9910(Fax), zaki@edge.jp

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Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

Hi,

I've configured both fr_timer and fr_inv_timer, like as followings.

loadmodule "/usr/local/lib/ser/modules/tm.so" modparam("tm", "fr_inv_timer", 30) modparam("tm", "fr_timer", 20)

If in the case of an endpoint is dead, INVITE request retransmission must be end after 20seconds with above configuration?

Best Regards,

At 06:29 PM 12/9/2003, Alan Crosswell wrote:

...

I've had this same problem, trying to do forwarding to voicemail. If the voicemail does not answer (using a separate vm server -- asterisk) then the invite keeps going forever. In fact, the sip phone rings forever even after the original inviter has canceled.

Sounds like a possible misconfiguration to me.

If the destination is dead, SER gives up on retransmission (fr_timer). If the destination does not send a final reply, SER cancels at fr_inv_timer.

-jiri

...

I didn't add the append_branch("sip:parallel@localhost:9"). Is that necessary for some reason?

It is necessary if you wish to forward to this address. I suppose you don't want to forward to a sink destination and this is just a value taken from an example.